[Oct 10 13:52:22] Asterisk SVN-branch-12-r400824 built by root @ on a i686 running Linux on 2013-10-01 16:12:22 UTC [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DBGet [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DBPut [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DBDel [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DBDelTree [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'MESSAGE' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'MESSAGE_DATA' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'MessageSend' [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action MessageSend [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DataGet [Oct 10 13:52:22] VERBOSE[15559] channel.c: == Registered channel type 'Surrogate' (Surrogate channel used to pull channel from an application) [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/codecs.conf': Found [Oct 10 13:52:22] VERBOSE[15559] loader.c: Asterisk Dynamic Loader Starting: [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/modules.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/dnsmgr.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/acl.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/http.conf': Found [Oct 10 13:52:22] VERBOSE[15559] http.c: Bound HTTP server to address 0.0.0.0:0 [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/indications.conf': Found [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'at' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'au' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'bg' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'br' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'be' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'ch' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'cl' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'cn' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'cz' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'de' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'dk' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'ee' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'es' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'fi' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'fr' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'gr' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'hu' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'il' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'in' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'it' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'lt' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'jp' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'mx' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'my' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'nl' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'no' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'nz' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'ph' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'pl' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'pt' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'ru' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'se' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'sg' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'th' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'uk' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'us' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'us-old' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'tw' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 've' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Registered indication country 'za' [Oct 10 13:52:22] VERBOSE[15559] indications.c: -- Setting default indication country to 'us' [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/features.conf': Found [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'FEATURE' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'FEATUREMAP' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Bridge' [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Bridge [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr.conf': Found [Oct 10 13:52:22] NOTICE[15559] cdr.c: CDR simple logging enabled. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/dsp.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/udptl.conf': Found [Oct 10 13:52:22] VERBOSE[15559] pbx.c: Asterisk PBX Core Initializing [Oct 10 13:52:22] VERBOSE[15559] pbx.c: Registering builtin applications: [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'EXCEPTION' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'TESTTIME' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Answer] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Answer' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [BackGround] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'BackGround' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Busy] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Busy' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Congestion] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Congestion' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [ExecIfTime] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'ExecIfTime' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Goto] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Goto' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [GotoIf] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'GotoIf' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [GotoIfTime] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'GotoIfTime' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [ImportVar] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'ImportVar' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Hangup] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Hangup' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Incomplete] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Incomplete' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [NoOp] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'NoOp' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Proceeding] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Proceeding' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Progress] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Progress' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [RaiseException] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'RaiseException' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Ringing] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Ringing' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [SayAlpha] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SayAlpha' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [SayAlphaCase] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SayAlphaCase' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [SayDigits] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SayDigits' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [SayNumber] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SayNumber' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [SayPhonetic] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SayPhonetic' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Set] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Set' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [MSet] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'MSet' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [SetAMAFlags] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SetAMAFlags' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [Wait] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Wait' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: [WaitExten] [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'WaitExten' [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action ShowDialPlan [Oct 10 13:52:22] VERBOSE[15559] channel.c: == Registered channel type 'Local' (Local Proxy Channel Driver) [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action LocalOptimizeAway [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cel.conf': Found [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Ping [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Events [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Logoff [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Login [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Challenge [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Hangup [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Status [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Setvar [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Getvar [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action GetConfig [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action GetConfigJSON [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action UpdateConfig [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action CreateConfig [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action ListCategories [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Redirect [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Atxfer [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Originate [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Command [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action ExtensionState [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action PresenceState [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action AbsoluteTimeout [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action MailboxStatus [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action MailboxCount [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action ListCommands [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SendText [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action UserEvent [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action WaitEvent [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action CoreSettings [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action CoreStatus [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Reload [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action CoreShowChannels [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action ModuleLoad [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action ModuleCheck [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action AOCMessage [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Filter [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action BlindTransfer [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'AMI_CLIENT' [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/manager.conf': Found [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action BridgeList [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action BridgeInfo [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/users.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/enum.conf': Found [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'CallCompletionRequest' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'CallCompletionCancel' [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/ccss.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/ccss.conf': Found [Oct 10 13:52:22] VERBOSE[15559] loader.c: Asterisk Dynamic Loader Starting: [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/modules.conf': Found [Oct 10 13:52:22] NOTICE[15559] loader.c: 281 modules will be loaded. [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_statsd.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/statsd.conf': Found [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_statsd.so => (Statsd client support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_odbc.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/res_odbc.conf': Found [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'ODBC_Commit' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'ODBC_Rollback' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'ODBC' [Oct 10 13:52:22] NOTICE[15559] res_odbc.c: res_odbc loaded. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_odbc.so => (ODBC resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_sorcery_config.so. [Oct 10 13:52:22] VERBOSE[15559] sorcery.c: == Sorcery registered wizard 'config' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_sorcery_config.so => (Sorcery Configuration File Object Wizard) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_sorcery_astdb.so. [Oct 10 13:52:22] VERBOSE[15559] sorcery.c: == Sorcery registered wizard 'astdb' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_sorcery_astdb.so => (Sorcery Astdb Object Wizard) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_sorcery_memory.so. [Oct 10 13:52:22] VERBOSE[15559] sorcery.c: == Sorcery registered wizard 'memory' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_sorcery_memory.so => (Sorcery In-Memory Object Wizard) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_sorcery_realtime.so. [Oct 10 13:52:22] VERBOSE[15559] sorcery.c: == Sorcery registered wizard 'realtime' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_sorcery_realtime.so => (Sorcery Realtime Object Wizard) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_log_forwarder.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_log_forwarder.so => (PJSIP Log Forwarder) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/sorcery.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/pjsip.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/sorcery.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/pjsip.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/pjsip.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/pjsip.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/pjsip.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/pjsip.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/pjsip.conf': Found [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action PJSIPQualify [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip.so => (Basic SIP resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_http_websocket.so. [Oct 10 13:52:22] VERBOSE[15559] res_http_websocket.c: == WebSocket registered sub-protocol 'echo' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_http_websocket.so => (HTTP WebSocket Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_srtp.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_srtp.so => (Secure RTP (SRTP)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_stun_monitor.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/res_stun_monitor.conf': Found [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_stun_monitor.so => (STUN Network Monitor) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_smdi.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/smdi.conf': Found [Oct 10 13:52:22] NOTICE[15559] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_crypto.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_crypto.so => (Cryptographic Digital Signatures) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_pubsub.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_pubsub.so => (PJSIP event resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_exten_state.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_exten_state.so => (PJSIP Extension State Notifications) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_monitor.so. [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Monitor' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'StopMonitor' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'ChangeMonitor' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'PauseMonitor' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'UnpauseMonitor' [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Monitor [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action StopMonitor [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action ChangeMonitor [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action PauseMonitor [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action UnpauseMonitor [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_monitor.so => (Call Monitoring Resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_xmpp.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/xmpp.conf': Found [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action JabberSend [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'JabberSend' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'JabberSendGroup' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'JabberStatus' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'JabberJoin' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'JabberLeave' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'JABBER_STATUS' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'JABBER_RECEIVE' [Oct 10 13:52:22] VERBOSE[15559] message.c: -- Message technology handler 'xmpp' registered. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_xmpp.so => (Asterisk XMPP Interface) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_stasis_recording.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_stasis_recording.so => (Stasis application recording support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_ari_model.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_ari_model.so => (ARI Model validators) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_agi.so. [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'answer' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'asyncagi break' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'channel status' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'database del' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'database deltree' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'database get' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'database put' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'exec' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'get data' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'get full variable' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'get option' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'get variable' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'hangup' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'noop' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'receive char' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'receive text' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'record file' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'say alpha' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'say digits' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'say number' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'say phonetic' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'say date' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'say time' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'say datetime' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'send image' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'send text' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'set autohangup' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'set callerid' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'set context' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'set extension' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'set music' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'set priority' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'set variable' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'stream file' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'control stream file' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'tdd mode' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'verbose' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'wait for digit' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'speech create' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'speech set' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'speech destroy' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'speech load grammar' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'speech unload grammar' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'speech activate grammar' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'speech deactivate grammar' registered [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'speech recognize' registered [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'DeadAGI' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'EAGI' [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action AGI [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'AGI' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_agi.so => (Asterisk Gateway Interface (AGI)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_speech.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_speech.so => (Generic Speech Recognition API) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_fax.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/res_fax.conf': Found [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SendFAX' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'ReceiveFAX' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'FAXOPT' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_fax.so => (Generic FAX Applications) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_session.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_session.so => (PJSIP Session resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_ari.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/ari.conf': Found [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_ari.so => (Asterisk RESTful Interface) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_calendar.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/calendar.conf': Found [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'CALENDAR_BUSY' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'CALENDAR_EVENT' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'CALENDAR_QUERY' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'CALENDAR_QUERY_RESULT' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'CALENDAR_WRITE' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_calendar.so => (Asterisk Calendar integration) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_stasis.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_stasis.so => (Stasis application support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_stasis_answer.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_stasis_answer.so => (Stasis application answer support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_ael_share.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_ael_share.so => (share-able code for AEL) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_stasis_playback.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_stasis_playback.so => (Stasis application playback support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading pbx_lua.so. [Oct 10 13:52:22] ERROR[15559] pbx_lua.c: Error loading extensions.lua: Unable to find 'extensions' table in extensions.lua [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_parking.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/res_parking.conf': Found [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Registered extension context 'parkedcalls'; registrar: res_parking [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '700' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '701' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '702' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '703' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '704' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '705' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '706' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '707' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '708' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '709' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '710' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '711' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '712' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '713' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '714' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '715' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '716' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '717' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '718' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '719' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: -- Added extension '720' priority 1 to parkedcalls [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Park' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'ParkedCall' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'ParkAndAnnounce' [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Parkinglots [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action ParkedCalls [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Park [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_parking.so => (Call Parking Resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_curl.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_curl.so => (cURL Resource Module) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading func_curl.so. [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'CURL' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'CURLOPT' [Oct 10 13:52:22] VERBOSE[15559] loader.c: func_curl.so => (Load external URL) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_config_sqlite3.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/res_config_sqlite3.conf': Found [Oct 10 13:52:22] NOTICE[15559] config.c: Registered Config Engine sqlite3 [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_config_sqlite3.so => (SQLite 3 realtime config engine) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_config_odbc.so. [Oct 10 13:52:22] NOTICE[15559] config.c: Registered Config Engine odbc [Oct 10 13:52:22] VERBOSE[15559] res_config_odbc.c: res_config_odbc loaded. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_config_odbc.so => (Realtime ODBC configuration) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_config_ldap.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/res_ldap.conf': Found [Oct 10 13:52:22] NOTICE[15559] res_config_ldap.c: No directory user found, anonymous binding as default. [Oct 10 13:52:22] ERROR[15559] res_config_ldap.c: No directory URL or host found. [Oct 10 13:52:22] ERROR[15559] res_config_ldap.c: Cannot load LDAP RealTime driver. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_config_ldap.so => (LDAP realtime interface) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_config_pgsql.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/res_pgsql.conf': Found [Oct 10 13:52:22] ERROR[15559] res_config_pgsql.c: PostgreSQL RealTime: Failed to connect database asterisk on 127.0.0.1: [Oct 10 13:52:22] WARNING[15559] res_config_pgsql.c: PostgreSQL RealTime: Couldn't establish connection. Check debug. [Oct 10 13:52:22] VERBOSE[15559] res_config_pgsql.c: == PostgreSQL RealTime reloaded. [Oct 10 13:52:22] NOTICE[15559] config.c: Registered Config Engine pgsql [Oct 10 13:52:22] VERBOSE[15559] res_config_pgsql.c: PostgreSQL RealTime driver loaded. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_config_pgsql.so => (PostgreSQL RealTime Configuration Driver) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_config_curl.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/res_curl.conf': Found [Oct 10 13:52:22] NOTICE[15559] config.c: Registered Config Engine curl [Oct 10 13:52:22] VERBOSE[15559] res_config_curl.c: res_config_curl loaded. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_config_curl.so => (Realtime Curl configuration) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_config_sqlite.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/res_config_sqlite.conf': Found [Oct 10 13:52:22] NOTICE[15559] config.c: Registered Config Engine sqlite [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_config_sqlite.so => (Realtime SQLite configuration) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_timing_timerfd.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_timing_timerfd.so => (Timerfd Timing Interface) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_timing_pthread.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_timing_pthread.so => (pthread Timing Interface) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_timing_dahdi.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_timing_dahdi.so => (DAHDI Timing Interface) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_musiconhold.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/musiconhold.conf': Found [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'MusicOnHold' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'WaitMusicOnHold' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SetMusicOnHold' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'StartMusicOnHold' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'StopMusicOnHold' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_musiconhold.so => (Music On Hold Resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_rtp_asterisk.so. [Oct 10 13:52:22] VERBOSE[15559] rtp_engine.c: == Registered RTP engine 'asterisk' [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/rtp.conf': Found [Oct 10 13:52:22] VERBOSE[15559] res_rtp_asterisk.c: == RTP Allocating from port range 10000 -> 20000 [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_rtp_asterisk.so => (Asterisk RTP Stack) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_authenticator_digest.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_authenticator_digest.so => (PJSIP authentication resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_pidf.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_pidf.so => (PJSIP Extension State PIDF Provider) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_mwi.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_mwi.so => (PJSIP MWI resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_format_attr_silk.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/codecs.conf': Found [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_format_attr_silk.so => (SILK Format Attribute Module) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_outbound_authenticator_digest.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_outbound_authenticator_digest.so => (PJSIP authentication resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_format_attr_celt.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/codecs.conf': Found [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_format_attr_celt.so => (CELT Format Attribute Module) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_rtp_multicast.so. [Oct 10 13:52:22] VERBOSE[15559] rtp_engine.c: == Registered RTP engine 'multicast' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_rtp_multicast.so => (Multicast RTP Engine) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_format_attr_opus.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/codecs.conf': Found [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_format_attr_opus.so => (Opus Format Attribute Module) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_sdp_rtp.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_sdp_rtp.so => (PJSIP SDP RTP/AVP stream handler) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading chan_mgcp.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/mgcp.conf': Found [Oct 10 13:52:22] VERBOSE[15559] chan_mgcp.c: == MGCP Listening on 0.0.0.0:2727 [Oct 10 13:52:22] VERBOSE[15559] channel.c: == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [Oct 10 13:52:22] VERBOSE[15559] rtp_engine.c: == Registered RTP glue 'MGCP' [Oct 10 13:52:22] VERBOSE[15559] loader.c: chan_mgcp.so => (Media Gateway Control Protocol (MGCP)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading chan_dahdi.so. [Oct 10 13:52:22] VERBOSE[15559] bridge.c: == Registered bridge technology native_dahdi [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'DAHDISendKeypadFacility' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'DAHDISendCallreroutingFacility' [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/chan_dahdi.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/users.conf': Found [Oct 10 13:52:22] VERBOSE[15559] chan_dahdi.c: -- Automatically generated pseudo channel [Oct 10 13:52:22] WARNING[15559] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23. [Oct 10 13:52:22] WARNING[15559] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Oct 10 13:52:22] WARNING[15559] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35. [Oct 10 13:52:22] WARNING[15559] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39. [Oct 10 13:52:22] WARNING[15559] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Oct 10 13:52:22] VERBOSE[15559] channel.c: == Registered channel type 'DAHDI' (DAHDI Telephony w/PRI) [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DAHDITransfer [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DAHDIHangup [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DAHDIDialOffhook [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DAHDIDNDon [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DAHDIDNDoff [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DAHDIShowChannels [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action DAHDIRestart [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action PRIShowSpans [Oct 10 13:52:22] VERBOSE[15559] loader.c: chan_dahdi.so => (DAHDI Telephony w/PRI) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading chan_motif.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/motif.conf': Found [Oct 10 13:52:22] VERBOSE[15559] rtp_engine.c: == Registered RTP glue 'Motif' [Oct 10 13:52:22] VERBOSE[15559] channel.c: == Registered channel type 'Motif' (Motif Jingle Channel Driver) [Oct 10 13:52:22] VERBOSE[15559] loader.c: chan_motif.so => (Motif Jingle Channel Driver) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading chan_multicast_rtp.so. [Oct 10 13:52:22] VERBOSE[15559] channel.c: == Registered channel type 'MulticastRTP' (Multicast RTP Paging Channel Driver) [Oct 10 13:52:22] VERBOSE[15559] loader.c: chan_multicast_rtp.so => (Multicast RTP Paging Channel) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading chan_iax2.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/iax.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/users.conf': Found [Oct 10 13:52:22] VERBOSE[15559] chan_iax2.c: == Binding IAX2 to address 0.0.0.0:4569 [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'IAX2Provision' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'IAXPEER' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'IAXVAR' [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action IAXpeers [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action IAXpeerlist [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action IAXnetstats [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action IAXregistry [Oct 10 13:52:22] VERBOSE[15559] channel.c: == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) [Oct 10 13:52:22] VERBOSE[15559] chan_iax2.c: == 10 helper threads started [Oct 10 13:52:22] VERBOSE[15559] chan_iax2.c: == IAX Ready and Listening [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/iaxprov.conf': Found [Oct 10 13:52:22] VERBOSE[15559] iax2/provision.c: -- Loaded provisioning template 'default' [Oct 10 13:52:22] VERBOSE[15559] loader.c: chan_iax2.so => (Inter Asterisk eXchange (Ver 2)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading chan_skinny.so. [Oct 10 13:52:22] NOTICE[15559] chan_skinny.c: Configuring skinny from skinny.conf [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/skinny.conf': Found [Oct 10 13:52:22] VERBOSE[15559] chan_skinny.c: == Skinny listening on 0.0.0.0:2000 [Oct 10 13:52:22] VERBOSE[15559] channel.c: == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [Oct 10 13:52:22] VERBOSE[15559] rtp_engine.c: == Registered RTP glue 'Skinny' [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SKINNYdevices [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SKINNYshowdevice [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SKINNYlines [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SKINNYshowline [Oct 10 13:52:22] VERBOSE[15559] loader.c: chan_skinny.so => (Skinny Client Control Protocol (Skinny)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading chan_sip.so. [Oct 10 13:52:22] VERBOSE[15559] chan_sip.c: SIP channel loading... [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/sip.conf': Found [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/users.conf': Found [Oct 10 13:52:22] VERBOSE[15559] chan_sip.c: == SIP Listening on 0.0.0.0:5061 [Oct 10 13:52:22] VERBOSE[15559] netsock2.c: == Using SIP CoS mark 4 [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/sip_notify.conf': Found [Oct 10 13:52:22] VERBOSE[15559] message.c: -- Message technology handler 'sip' registered. [Oct 10 13:52:22] VERBOSE[15559] channel.c: == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) [Oct 10 13:52:22] VERBOSE[15559] rtp_engine.c: == Registered RTP glue 'SIP' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SIPDtmfMode' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SIPAddHeader' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SIPRemoveHeader' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'SIPSendCustomINFO' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'SIP_HEADER' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'SIPPEER' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'SIPCHANINFO' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'CHECKSIPDOMAIN' [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SIPpeers [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SIPshowpeer [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SIPqualifypeer [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SIPshowregistry [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SIPnotify [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action SIPpeerstatus [Oct 10 13:52:22] VERBOSE[15559] res_http_websocket.c: == WebSocket registered sub-protocol 'sip' [Oct 10 13:52:22] VERBOSE[15559] loader.c: chan_sip.so => (Session Initiation Protocol (SIP)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading chan_pjsip.so. [Oct 10 13:52:22] VERBOSE[15559] rtp_engine.c: == Registered RTP glue 'PJSIP' [Oct 10 13:52:22] VERBOSE[15559] channel.c: == Registered channel type 'PJSIP' (PJSIP Channel Driver) [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'PJSIP_DIAL_CONTACTS' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'PJSIP_MEDIA_OFFER' [Oct 10 13:52:22] VERBOSE[15559] loader.c: chan_pjsip.so => (PJSIP Channel Driver) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_t38.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_t38.so => (PJSIP T.38 UDPTL Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_sln.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format sln, extension(s) sln|raw [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format sln12, extension(s) sln12 [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format sln16, extension(s) sln16 [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format sln24, extension(s) sln24 [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format sln32, extension(s) sln32 [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format sln44, extension(s) sln44 [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format sln48, extension(s) sln48 [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format sln96, extension(s) sln96 [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format sln192, extension(s) sln192 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_sln.so => (Raw Signed Linear Audio support (SLN) 8khz-192khz) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_acl.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_acl.so => (PJSIP ACL Resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_caller_id.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_caller_id.so => (PJSIP Caller ID Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_siren14.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format siren14, extension(s) siren14 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_siren14.so => (ITU G.722.1 Annex C (Siren14, licensed from Polycom)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_wav_gsm.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format wav49, extension(s) WAV|wav49 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_messaging.so. [Oct 10 13:52:22] VERBOSE[15559] message.c: -- Message technology handler 'pjsip' registered. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_messaging.so => (PJSIP Messaging Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_rfc3326.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_rfc3326.so => (PJSIP RFC3326 Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_nat.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_nat.so => (PJSIP NAT Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_ilbc.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format iLBC, extension(s) ilbc [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_ilbc.so => (Raw iLBC data) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_g723.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format g723sf, extension(s) g723|g723sf [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_g723.so => (G.723.1 Simple Timestamp File Format) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_dtmf_info.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_dtmf_info.so => (PJSIP DTMF INFO Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_adsi.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/adsi.conf': Found [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_adsi.so => (ADSI Resource) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_header_funcs.so. [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'PJSIP_HEADER' [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_header_funcs.so => (PJSIP Header Functions) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_g729.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format g729, extension(s) g729 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_g729.so => (Raw G.729 data) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_h263.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format h263, extension(s) h263 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_h263.so => (Raw H.263 data) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_endpoint_identifier_user.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_endpoint_identifier_user.so => (PJSIP username endpoint identifier) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_wav.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format wav, extension(s) wav [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format wav16, extension(s) wav16 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_wav.so => (Microsoft WAV/WAV16 format (8kHz/16kHz Signed Linear)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading app_stack.so. [Oct 10 13:52:22] VERBOSE[15559] res_agi.c: == AGI Command 'gosub' registered [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'StackPop' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Return' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'GosubIf' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'Gosub' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'LOCAL' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'LOCAL_PEEK' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'STACK_PEEK' [Oct 10 13:52:22] VERBOSE[15559] loader.c: app_stack.so => (Dialplan subroutines (Gosub, Return, etc)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_diversion.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_diversion.so => (PJSIP Add Diversion Header Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_jpeg.so. [Oct 10 13:52:22] VERBOSE[15559] image.c: == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_jpeg.so => (jpeg (joint picture experts group) image format) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_siren7.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format siren7, extension(s) siren7 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_siren7.so => (ITU G.722.1 (Siren7, licensed from Polycom)) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_ogg_vorbis.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format ogg_vorbis, extension(s) ogg [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_ogg_vorbis.so => (OGG/Vorbis audio) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_transport_websocket.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_vox.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format vox, extension(s) vox [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_vox.so => (Dialogic VOX (ADPCM) File Format) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_outbound_registration.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/pjsip.conf': Found [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action PJSIPUnregister [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_outbound_registration.so => (PJSIP Outbound Registration Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_pcm.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format pcm, extension(s) pcm|ulaw|ul|mu|ulw [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format alaw, extension(s) alaw|al|alw [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format au, extension(s) au [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format g722, extension(s) g722 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_pcm.so => (Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G.722 16Khz) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_g726.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format g726-40, extension(s) g726-40 [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format g726-32, extension(s) g726-32 [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format g726-24, extension(s) g726-24 [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format g726-16, extension(s) g726-16 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_g726.so => (Raw G.726 (16/24/32/40kbps) data) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading func_dialplan.so. [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'DIALPLAN_EXISTS' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'VALID_EXTEN' [Oct 10 13:52:22] VERBOSE[15559] loader.c: func_dialplan.so => (Dialplan Context/Extension/Priority Checking Functions) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_notify.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/pjsip_notify.conf': Found [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action PJSIPNotify [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_notify.so => (CLI/AMI PJSIP NOTIFY Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_logger.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_logger.so => (PJSIP Packet Logger) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_gsm.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format gsm, extension(s) gsm [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_gsm.so => (Raw GSM data) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_endpoint_identifier_ip.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/pjsip.conf': Found [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_endpoint_identifier_ip.so => (PJSIP IP endpoint identifier) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_registrar.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_registrar.so => (PJSIP Registrar Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_one_touch_record_info.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_one_touch_record_info.so => (PJSIP INFO One Touch Recording Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_refer.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_refer.so => (PJSIP Blind and Attended Transfer Support) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading res_pjsip_registrar_expire.so. [Oct 10 13:52:22] VERBOSE[15559] loader.c: res_pjsip_registrar_expire.so => (PJSIP Contact Auto-Expiration) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_g719.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format g719, extension(s) g719 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_g719.so => (ITU G.719) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading format_h264.so. [Oct 10 13:52:22] VERBOSE[15559] file.c: == Registered file format h264, extension(s) h264 [Oct 10 13:52:22] VERBOSE[15559] loader.c: format_h264.so => (Raw H.264 data) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading func_presencestate.so. [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'PRESENCE_STATE' [Oct 10 13:52:22] VERBOSE[15559] loader.c: func_presencestate.so => (Gets or sets a presence state in the dialplan) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading app_agent_pool.so. [Oct 10 13:52:22] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/agents.conf': Found [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action Agents [Oct 10 13:52:22] VERBOSE[15559] manager.c: == Manager registered action AgentLogoff [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered custom function 'AGENT' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'AgentLogin' [Oct 10 13:52:22] VERBOSE[15559] pbx.c: == Registered application 'AgentRequest' [Oct 10 13:52:22] VERBOSE[15559] loader.c: app_agent_pool.so => (Call center agent pool applications) [Oct 10 13:52:22] VERBOSE[15559] loader.c: Loading app_confbridge.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/confbridge.conf': Found [Oct 10 13:52:23] VERBOSE[15559] channel.c: == Registered channel type 'CBRec' (Conference Bridge Recording Channel) [Oct 10 13:52:23] VERBOSE[15559] channel.c: == Registered channel type 'CBAnn' (Conference Bridge Announcing Channel) [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ConfBridge' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CONFBRIDGE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CONFBRIDGE_INFO' [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ConfbridgeList [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ConfbridgeListRooms [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ConfbridgeMute [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ConfbridgeUnmute [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ConfbridgeKick [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ConfbridgeUnlock [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ConfbridgeLock [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ConfbridgeStartRecord [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ConfbridgeStopRecord [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ConfbridgeSetSingleVideoSrc [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_confbridge.so => (Conference Bridge Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_devstate.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'DEVICE_STATE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'HINT' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_devstate.so => (Gets or sets a device state in the dialplan) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_calendar_icalendar.so. [Oct 10 13:52:23] VERBOSE[15559] res_calendar.c: == Registered calendar type 'ical' (iCalendar .ics calendars) [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_calendar_icalendar.so => (Asterisk iCalendar .ics file integration) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_calendar_exchange.so. [Oct 10 13:52:23] VERBOSE[15559] res_calendar.c: == Registered calendar type 'exchange' (MS Exchange calendars) [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_calendar_exchange.so => (Asterisk MS Exchange Calendar Integration) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_calendar_caldav.so. [Oct 10 13:52:23] VERBOSE[15559] res_calendar.c: == Registered calendar type 'caldav' (CalDAV calendars) [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_calendar_caldav.so => (Asterisk CalDAV Calendar Integration) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_calendar_ews.so. [Oct 10 13:52:23] VERBOSE[15559] res_calendar.c: == Registered calendar type 'ews' (MS Exchange Web Service calendars) [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_calendar_ews.so => (Asterisk MS Exchange Web Service Calendar Integration) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cdr_odbc.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr_odbc.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: cdr_odbc.so => (ODBC CDR Backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cdr_csv.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: cdr_csv.so => (Comma Separated Values CDR Backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cel_manager.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cel.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: cel_manager.so => (Asterisk Manager Interface CEL Backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cel_pgsql.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cel_pgsql.conf': Found [Oct 10 13:52:23] WARNING[15559] cel_pgsql.c: CEL pgsql config file missing global section. [Oct 10 13:52:23] VERBOSE[15559] loader.c: cel_pgsql.so => (PostgreSQL CEL Backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cdr_custom.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr_custom.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: cdr_custom.so => (Customizable Comma Separated Values CDR Backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cel_radius.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cel.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: cel_radius.so => (RADIUS CEL Backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cdr_radius.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: cdr_radius.so => (RADIUS CDR Backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cdr_sqlite3_custom.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr_sqlite3_custom.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cel_odbc.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cel_odbc.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: cel_odbc.so => (ODBC CEL backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cdr_pgsql.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr_pgsql.conf': Found [Oct 10 13:52:23] NOTICE[15559] cdr_pgsql.c: cdr_pgsql configuration contains no global section, skipping module load. [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cdr_adaptive_odbc.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr_adaptive_odbc.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: cdr_adaptive_odbc.so => (Adaptive ODBC CDR backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cdr_manager.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr_manager.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: cdr_manager.so => (Asterisk Manager Interface CDR Backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cel_custom.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cel_custom.conf': Found [Oct 10 13:52:23] NOTICE[15559] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. [Oct 10 13:52:23] VERBOSE[15559] cel_custom.c: Added CEL CSV mapping for 0 files. [Oct 10 13:52:23] VERBOSE[15559] loader.c: cel_custom.so => (Customizable Comma Separated Values CEL Backend) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cel_sqlite3_custom.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cel_sqlite3_custom.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cdr_tds.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr_tds.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cdr_syslog.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cdr_syslog.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading cel_tds.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cel_tds.conf': Found [Oct 10 13:52:23] NOTICE[15559] cel_tds.c: cel_tds has no global category, nothing to configure. [Oct 10 13:52:23] WARNING[15559] cel_tds.c: cel_tds module had config problems; declining load [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_alarmreceiver.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/alarmreceiver.conf': Found [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'AlarmReceiver' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_alarmreceiver.so => (Alarm Receiver for Asterisk) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_festival.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/festival.conf': Found [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Festival' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_festival.so => (Simple Festival Interface) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_clioriginate.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_clioriginate.so => (Call origination and redirection from the CLI) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_voicemail.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/voicemail.conf': Found [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/users.conf': Found [Oct 10 13:52:23] VERBOSE[15573] app.c: -- Message check requested for mailbox 1001@default/folder INBOX but voicemail not loaded. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'VoiceMail' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'VoiceMailMain' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MailboxExists' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'VMAuthenticate' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'VoiceMailPlayMsg' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'VMSayName' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'MAILBOX_EXISTS' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'VM_INFO' [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action VoicemailUsersList [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action VoicemailRefresh [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_voicemail.so => (Comedian Mail (Voicemail System)) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_srv.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SRVQUERY' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SRVRESULT' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_srv.so => (SRV related dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_frame_trace.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'FRAME_TRACE' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_frame_trace.so => (Frame Trace for internal ast_frame debugging.) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading chan_oss.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/oss.conf': Found [Oct 10 13:52:23] VERBOSE[15559] channel.c: == Registered channel type 'Console' (OSS Console Channel Driver) [Oct 10 13:52:23] VERBOSE[15559] loader.c: chan_oss.so => (OSS Console Channel Driver) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_while.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'While' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'EndWhile' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ExitWhile' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ContinueWhile' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_while.so => (While Loops and Conditional Execution) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_bridgewait.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'BridgeWait' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_bridgewait.so => (Place the channel into a holding bridge application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_groupcount.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'GROUP_COUNT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'GROUP_MATCH_COUNT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'GROUP_LIST' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'GROUP' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_groupcount.so => (Channel group dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_mixmonitor.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MixMonitor' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'StopMixMonitor' [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action MixMonitorMute [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action MixMonitor [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action StopMixMonitor [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_mixmonitor.so => (Mixed Audio Monitoring Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_pitchshift.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'PITCH_SHIFT' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_pitchshift.so => (Audio Effects Dialplan Functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_clialiases.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cli_aliases.conf': Found [Oct 10 13:52:23] VERBOSE[15559] res_clialiases.c: == Aliased CLI command 'hangup request' to 'channel request hangup' [Oct 10 13:52:23] VERBOSE[15559] res_clialiases.c: == Aliased CLI command 'originate' to 'channel originate' [Oct 10 13:52:23] VERBOSE[15559] res_clialiases.c: == Aliased CLI command 'help' to 'core show help' [Oct 10 13:52:23] VERBOSE[15559] res_clialiases.c: == Aliased CLI command 'pri intense debug span' to 'pri set debug intense span' [Oct 10 13:52:23] VERBOSE[15559] res_clialiases.c: == Aliased CLI command 'reload' to 'module reload' [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_clialiases.so => (CLI Aliases) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_controlplayback.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ControlPlayback' [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action ControlPlayback [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_controlplayback.so => (Control Playback Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading chan_bridge_media.so. [Oct 10 13:52:23] VERBOSE[15559] channel.c: == Registered channel type 'Announcer' (Bridge Media Announcing Channel Driver) [Oct 10 13:52:23] VERBOSE[15559] channel.c: == Registered channel type 'Recorder' (Bridge Media Recording Channel Driver) [Oct 10 13:52:23] VERBOSE[15559] loader.c: chan_bridge_media.so => (Bridge Media Channel Driver) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_convert.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_convert.so => (File format conversion CLI command) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading pbx_realtime.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: pbx_realtime.so => (Realtime Switch) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_timeout.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'TIMEOUT' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_timeout.so => (Channel timeout dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_global.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'GLOBAL' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SHARED' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_global.so => (Variable dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_module.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'IFMODULE' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_module.so => (Checks if Asterisk module is loaded in memory) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_amd.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/amd.conf': Found [Oct 10 13:52:23] VERBOSE[15559] app_amd.c: -- AMD defaults: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] maximumWordLength [5000] [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'AMD' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_amd.so => (Answering Machine Detection Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_cdr.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CDR' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CDR_PROP' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_cdr.so => (Call Detail Record (CDR) dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_mutestream.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'MUTEAUDIO' [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action MuteAudio [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_mutestream.so => (Mute audio stream resources) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_chanisavail.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ChanIsAvail' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_chanisavail.so => (Check channel availability) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_ilbc.so. [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'ilbctolin' from format ilbc to slin, table cost, 900000, computational cost 8001 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lintoilbc' from format slin to ilbc, table cost, 600000, computational cost 52002 [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_ilbc.so => (iLBC Coder/Decoder) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_dahdiras.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'DAHDIRAS' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_dahdiras.so => (DAHDI ISDN Remote Access Server) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_speech_utils.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SpeechCreate' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SpeechLoadGrammar' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SpeechUnloadGrammar' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SpeechActivateGrammar' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SpeechDeactivateGrammar' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SpeechStart' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SpeechBackground' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SpeechDestroy' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SpeechProcessingSound' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SPEECH' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SPEECH_SCORE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SPEECH_TEXT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SPEECH_GRAMMAR' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SPEECH_ENGINE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SPEECH_RESULTS_TYPE' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_speech_utils.so => (Dialplan Speech Applications) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_talkdetect.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'BackgroundDetect' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_talkdetect.so => (Playback with Talk Detection) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_sha1.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SHA1' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_sha1.so => (SHA-1 computation dialplan function) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_a_mu.so. [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'alawtoulaw' from format alaw to ulaw, table cost, 915000, computational cost 4001 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'ulawtoalaw' from format ulaw to alaw, table cost, 915000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_nbscat.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'NBScat' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_nbscat.so => (Silly NBS Stream Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_zapateller.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Zapateller' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_zapateller.so => (Block Telemarketers with Special Information Tone) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_sendtext.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SendText' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_sendtext.so => (Send Text Applications) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_channelredirect.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ChannelRedirect' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_channelredirect.so => (Redirects a given channel to a dialplan target) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading chan_phone.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/phone.conf': Found [Oct 10 13:52:23] VERBOSE[15559] channel.c: == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [Oct 10 13:52:23] VERBOSE[15559] loader.c: chan_phone.so => (Linux Telephony API Support) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_odbc.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ODBC_FETCH' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ODBCFinish' [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/func_odbc.conf': Found [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ODBC_SQL' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ODBC_ANTIGF' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ODBC_PRESENCE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SQL_ESC' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_odbc.so => (ODBC lookups) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_readexten.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ReadExten' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_readexten.so => (Read and evaluate extension validity) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_md5.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'MD5' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_md5.so => (MD5 digest dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_rand.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'RAND' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_rand.so => (Random number dialplan function) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_ari_endpoints.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_ari_endpoints.so => (RESTful API module - Endpoint resources) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading bridge_builtin_interval_features.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: bridge_builtin_interval_features.so => (Built in bridging interval features) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading pbx_config.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/extensions.conf': Found [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Registered extension context 'default'; registrar: pbx_config [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Registered extension context 'public'; registrar: pbx_config [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Registered extension context 'internal'; registrar: pbx_config [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1001' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1001' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1001' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1001' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1002' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1002' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1002' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1002' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1003' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1003' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1003' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1003' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '551' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '552' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '552' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '553' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '555' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '555' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'agent-1001' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '997' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '997' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 4 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 5 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 6 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 7 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '999' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '999' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '999' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '666' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'phone_A' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'phone_B' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '201' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '201' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '201' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/users.conf': Found [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Registered extension context 'parkedcalls'; registrar: res_parking/default [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '720' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '719' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '718' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '717' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '716' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '715' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '714' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '713' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '712' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '711' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '710' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '709' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '708' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '707' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '706' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '705' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '704' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '703' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '702' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '701' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '700' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.000316 sec [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Time to restore hints and swap in new dialplan: 0.000008 sec [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Time to delete the old dialplan: 0.000024 sec [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Total time merge_contexts_delete: 0.000348 sec [Oct 10 13:52:23] VERBOSE[15559] loader.c: pbx_config.so => (Text Extension Configuration) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_minivm.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MinivmRecord' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MinivmGreet' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MinivmNotify' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MinivmDelete' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MinivmAccMess' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MinivmMWI' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'MINIVMACCOUNT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'MINIVMCOUNTER' [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/minivm.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_minivm.so => (Mini VoiceMail (A minimal Voicemail e-mail System)) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_snmp.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/res_snmp.conf': Found [Oct 10 13:52:23] VERBOSE[15559] res_snmp.c: Loading [Sub]Agent Module [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_snmp.so => (SNMP [Sub]Agent for Asterisk) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_morsecode.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Morsecode' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_morsecode.so => (Morse code) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_math.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'MATH' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'INC' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'DEC' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_math.so => (Mathematical dialplan function) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_waitforring.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'WaitForRing' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_waitforring.so => (Waits until first ring after time) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading pbx_dundi.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/dundi.conf': Found [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'DUNDILOOKUP' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'DUNDIQUERY' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'DUNDIRESULT' [Oct 10 13:52:23] VERBOSE[15559] pbx_dundi.c: == DUNDi Ready and Listening on 0.0.0.0 port 4520 [Oct 10 13:52:23] VERBOSE[15559] loader.c: pbx_dundi.so => (Distributed Universal Number Discovery (DUNDi)) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_waitforsilence.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'WaitForSilence' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'WaitForNoise' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_waitforsilence.so => (Wait For Silence) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_aes.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'AES_DECRYPT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'AES_ENCRYPT' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_aes.so => (AES dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading chan_unistim.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/unistim.conf': Found [Oct 10 13:52:23] VERBOSE[15559] chan_unistim.c: == UNISTIM Listening on 0.0.0.0:5000 [Oct 10 13:52:23] VERBOSE[15559] channel.c: == Registered channel type 'USTM' (UNISTIM Channel Driver) [Oct 10 13:52:23] VERBOSE[15559] rtp_engine.c: == Registered RTP glue 'USTM' [Oct 10 13:52:23] VERBOSE[15559] loader.c: chan_unistim.so => (UNISTIM Protocol (USTM)) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading bridge_builtin_features.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: bridge_builtin_features.so => (Built in bridging features) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_limit.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_limit.so => (Resource limits) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_shell.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SHELL' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_shell.so => (Collects the output generated by a command executed by the system shell) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_originate.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Originate' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_originate.so => (Originate call) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_realtime.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_realtime.so => (Realtime Data Lookup/Rewrite) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_system.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'TrySystem' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'System' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_system.so => (Generic System() application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_extstate.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'EXTENSION_STATE' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_extstate.so => (Gets an extension's state in the dialplan) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_base64.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'BASE64_ENCODE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'BASE64_DECODE' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_base64.so => (base64 encode/decode dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_alaw.so. [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'alawtolin' from format alaw to slin, table cost, 900000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lintoalaw' from format slin to alaw, table cost, 600000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_alaw.so => (A-law Coder/Decoder) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_externalivr.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ExternalIVR' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_externalivr.so => (External IVR Interface Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_authenticate.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Authenticate' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_authenticate.so => (Authentication Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_volume.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'VOLUME' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_volume.so => (Technology independent volume control) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_macro.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MacroExit' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MacroIf' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MacroExclusive' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Macro' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_macro.so => (Extension Macros) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_sysinfo.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SYSINFO' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_sysinfo.so => (System information related functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_verbose.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Log' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Verbose' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_verbose.so => (Send verbose output) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_pjsip_endpoint_identifier_anonymous.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_pjsip_endpoint_identifier_anonymous.so => (PJSIP Anonymous endpoint identifier) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_format_attr_h264.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/codecs.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_format_attr_h264.so => (H.264 Format Attribute Module) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_stasis.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Stasis' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_stasis.so => (Stasis dialplan application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_ulaw.so. [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'ulawtolin' from format ulaw to slin, table cost, 900000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lintoulaw' from format slin to ulaw, table cost, 600000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lintotestlaw' from format slin to testlaw, table cost, 600000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'testlawtolin' from format testlaw to slin, table cost, 900000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_ulaw.so => (mu-Law Coder/Decoder) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_env.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ENV' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'STAT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'FILE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'FILE_COUNT_LINE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'FILE_FORMAT' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_env.so => (Environment/filesystem dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_cdr.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'NoCDR' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ResetCDR' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_milliwatt.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Milliwatt' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_milliwatt.so => (Digital Milliwatt (mu-law) Test Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_uri.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'URIDECODE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'URIENCODE' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_uri.so => (URI encode/decode dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading pbx_spool.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: pbx_spool.so => (Outgoing Spool Support) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_ari_channels.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_ari_channels.so => (RESTful API module - Channel resources) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_security_log.so. [Oct 10 13:52:23] VERBOSE[15559] res_security_log.c: -- Security Logging Enabled [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_security_log.so => (Security Event Logging) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_forkcdr.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ForkCDR' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_forkcdr.so => (Fork The CDR into 2 separate entities) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_jitterbuffer.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'JITTERBUFFER' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_jitterbuffer.so => (Jitter buffer for read side of channel.) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_userevent.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'UserEvent' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_userevent.so => (Custom User Event Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_sayunixtime.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SayUnixTime' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'DateTime' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_sayunixtime.so => (Say time) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_transfer.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Transfer' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_transfer.so => (Transfers a caller to another extension) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_dialgroup.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'DIALGROUP' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_dialgroup.so => (Dialgroup dialplan function) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading bridge_softmix.so. [Oct 10 13:52:23] VERBOSE[15559] bridge.c: == Registered bridge technology softmix [Oct 10 13:52:23] VERBOSE[15559] loader.c: bridge_softmix.so => (Multi-party software based channel mixing) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_chanspy.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ChanSpy' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ExtenSpy' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'DAHDIScan' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_chanspy.so => (Listen to the audio of an active channel) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_db.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'DBdel' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'DBdeltree' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_db.so => (Database Access Functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_playtones.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'PlayTones' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'StopPlayTones' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_playtones.so => (Playtones Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_logic.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ISNULL' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SET' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'EXISTS' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'IF' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'IFTIME' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'IMPORT' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_logic.so => (Logical dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_audiohookinherit.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'AUDIOHOOK_INHERIT' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_audiohookinherit.so => (Audiohook inheritance function) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_playback.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/say.conf': Found [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Playback' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_playback.so => (Sound File Playback Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_gsm.so. [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'gsmtolin' from format gsm to slin, table cost, 900000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lintogsm' from format slin to gsm, table cost, 600000, computational cost 4000 [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_gsm.so => (GSM Coder/Decoder) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading bridge_native_rtp.so. [Oct 10 13:52:23] VERBOSE[15559] bridge.c: == Registered bridge technology native_rtp [Oct 10 13:52:23] VERBOSE[15559] loader.c: bridge_native_rtp.so => (Native RTP bridging module) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_http_post.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/http.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_http_post.so => (HTTP POST support) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_ices.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ICES' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_ices.so => (Encode and Stream via icecast and ices) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_directory.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Directory' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_directory.so => (Extension Directory) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_iconv.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ICONV' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_iconv.so => (Charset conversions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_config.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'AST_CONFIG' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_config.so => (Asterisk configuration file variable access) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_sprintf.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SPRINTF' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_sprintf.so => (SPRINTF dialplan function) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_cut.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CUT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SORT' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_cut.so => (Cut out information from a string) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_dictate.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Dictate' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_dictate.so => (Virtual Dictation Machine) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_directed_pickup.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Pickup' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'PickupChan' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_directed_pickup.so => (Directed Call Pickup Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_adpcm.so. [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'adpcmtolin' from format adpcm to slin, table cost, 900000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lintoadpcm' from format slin to adpcm, table cost, 600000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_callcompletion.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CALLCOMPLETION' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_callcompletion.so => (Call Control Configuration Function) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_dumpchan.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'DumpChan' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_dumpchan.so => (Dump Info About The Calling Channel) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading bridge_holding.so. [Oct 10 13:52:23] VERBOSE[15559] bridge.c: == Registered bridge technology holding_bridge [Oct 10 13:52:23] VERBOSE[15559] loader.c: bridge_holding.so => (Holding bridge module) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_followme.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/followme.conf': Found [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'FollowMe' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_followme.so => (Find-Me/Follow-Me Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_g726.so. [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'g726tolin' from format g726 to slin, table cost, 900000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lintog726' from format slin to g726, table cost, 600000, computational cost 8001 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'g726aal2tolin' from format g726aal2 to slin, table cost, 900000, computational cost 1 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lintog726aal2' from format slin to g726aal2, table cost, 600000, computational cost 4000 [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_g726.so => (ITU G.726-32kbps G726 Transcoder) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_ari_events.so. [Oct 10 13:52:23] VERBOSE[15559] res_http_websocket.c: == WebSocket registered sub-protocol 'ari' [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_ari_events.so => (RESTful API module - WebSocket resource) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_g722.so. [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'g722tolin' from format g722 to slin, table cost, 960000, computational cost 4000 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lintog722' from format slin to g722, table cost, 825000, computational cost 4001 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'g722tolin16' from format g722 to slin16, table cost, 900000, computational cost 8000 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lin16tog722' from format slin16 to g722, table cost, 600000, computational cost 12001 [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_g722.so => (ITU G.722-64kbps G722 Transcoder) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_echo.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Echo' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_echo.so => (Simple Echo Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_waituntil.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'WaitUntil' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_waituntil.so => (Wait until specified time) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_privacy.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'PrivacyManager' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_privacy.so => (Require phone number to be entered, if no CallerID sent) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading bridge_simple.so. [Oct 10 13:52:23] VERBOSE[15559] bridge.c: == Registered bridge technology simple_bridge [Oct 10 13:52:23] VERBOSE[15559] loader.c: bridge_simple.so => (Simple two channel bridging module) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_page.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Page' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_page.so => (Page Multiple Phones) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_image.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SendImage' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_image.so => (Image Transmission Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_ari_recordings.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_ari_recordings.so => (RESTful API module - Recording resources) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_ari_playback.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_ari_playback.so => (RESTful API module - Playback control resources) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_callerid.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CALLERPRES' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CALLERID' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CONNECTEDLINE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'REDIRECTING' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_callerid.so => (Party ID related dialplan functions (Caller-ID, Connected-line, Redirecting)) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_adsiprog.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ADSIProg' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_adsiprog.so => (Asterisk ADSI Programming Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_softhangup.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SoftHangup' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_softhangup.so => (Hangs up the requested channel) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_senddtmf.so. [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action PlayDTMF [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SendDTMF' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_senddtmf.so => (Send DTMF digits Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_realtime.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'REALTIME' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'REALTIME_STORE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'REALTIME_DESTROY' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'REALTIME_FIELD' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'REALTIME_HASH' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_realtime.so => (Read/Write/Store/Destroy values from a RealTime repository) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_url.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SendURL' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_url.so => (Send URL Applications) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_strings.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'FIELDQTY' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'FIELDNUM' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'FILTER' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'REPLACE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'STRREPLACE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'LISTFILTER' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'REGEX' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ARRAY' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'QUOTE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CSV_QUOTE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'LEN' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'STRFTIME' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'STRPTIME' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'EVAL' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'KEYPADHASH' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'HASHKEYS' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'HASH' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ClearHash' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'TOUPPER' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'TOLOWER' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'SHIFT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'POP' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'PUSH' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'UNSHIFT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'PASSTHRU' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_strings.so => (String handling dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_mp3.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'MP3Player' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_mp3.so => (Silly MP3 Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_hangupcause.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'HANGUPCAUSE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'HANGUPCAUSE_KEYS' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'HangupCauseClear' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_hangupcause.so => (HANGUPCAUSE related functions and applications) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_dahdi.so. [Oct 10 13:52:23] ERROR[15559] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_dahdi.so => (Generic DAHDI Transcoder Codec Translator) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_ari_applications.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_ari_applications.so => (RESTful API module - Stasis application resources) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_vmcount.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'VMCOUNT' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_vmcount.so => (Indicator for whether a voice mailbox has messages in a given folder.) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_phoneprov.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'PP_EACH_USER' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'PP_EACH_EXTENSION' [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/sip.conf': Found [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/users.conf': Found [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/phoneprov.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_phoneprov.so => (HTTP Phone Provisioning) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_sms.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'SMS' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_sms.so => (SMS/PSTN handler) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_enum.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ENUMRESULT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ENUMQUERY' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'ENUMLOOKUP' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'TXTCIDNAME' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_enum.so => (ENUM related dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_version.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'VERSION' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_version.so => (Get Asterisk Version/Build Info) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_blacklist.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'BLACKLIST' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_blacklist.so => (Look up Caller*ID name/number from blacklist database) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_ari_bridges.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_ari_bridges.so => (RESTful API module - Bridge resources) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_test.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'TestClient' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'TestServer' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_test.so => (Interface Test Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_format_attr_h263.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/codecs.conf': Found [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/codecs.conf': Found [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_format_attr_h263.so => (H.263 Format Attribute Module) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_record.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Record' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_record.so => (Trivial Record Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_getcpeid.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'GetCPEID' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_getcpeid.so => (Get ADSI CPE ID) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_db.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'DB' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'DB_EXISTS' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'DB_DELETE' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'DB_KEYS' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_db.so => (Database (astdb) related dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_lpc10.so. [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lpc10tolin' from format lpc10 to slin, table cost, 900000, computational cost 4000 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'lintolpc10' from format slin to lpc10, table cost, 600000, computational cost 4000 [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_lpc10.so => (LPC10 2.4kbps Coder/Decoder) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_read.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Read' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_read.so => (Read Variable Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_ari_sounds.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_ari_sounds.so => (RESTful API module - Sound resources) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_dial.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Dial' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'RetryDial' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_dial.so => (Dialing Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_ari_asterisk.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_ari_asterisk.so => (RESTful API module - Asterisk resources) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading codec_resample.so. [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 8000khz -> 12000khz' from format slin to slin12, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 8000khz -> 16000khz' from format slin to slin16, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 8000khz -> 24000khz' from format slin to slin24, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 8000khz -> 32000khz' from format slin to slin32, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 8000khz -> 44100khz' from format slin to slin44, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 8000khz -> 48000khz' from format slin to slin48, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 8000khz -> 96000khz' from format slin to slin96, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 8000khz -> 192000khz' from format slin to slin192, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 12000khz -> 8000khz' from format slin12 to slin, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 12000khz -> 16000khz' from format slin12 to slin16, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 12000khz -> 24000khz' from format slin12 to slin24, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 12000khz -> 32000khz' from format slin12 to slin32, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 12000khz -> 44100khz' from format slin12 to slin44, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 12000khz -> 48000khz' from format slin12 to slin48, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 12000khz -> 96000khz' from format slin12 to slin96, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 12000khz -> 192000khz' from format slin12 to slin192, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 16000khz -> 8000khz' from format slin16 to slin, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 16000khz -> 12000khz' from format slin16 to slin12, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 16000khz -> 24000khz' from format slin16 to slin24, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 16000khz -> 32000khz' from format slin16 to slin32, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 16000khz -> 44100khz' from format slin16 to slin44, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 16000khz -> 48000khz' from format slin16 to slin48, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 16000khz -> 96000khz' from format slin16 to slin96, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 16000khz -> 192000khz' from format slin16 to slin192, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 24000khz -> 8000khz' from format slin24 to slin, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 24000khz -> 12000khz' from format slin24 to slin12, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 24000khz -> 16000khz' from format slin24 to slin16, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 24000khz -> 32000khz' from format slin24 to slin32, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 24000khz -> 44100khz' from format slin24 to slin44, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 24000khz -> 48000khz' from format slin24 to slin48, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 24000khz -> 96000khz' from format slin24 to slin96, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 24000khz -> 192000khz' from format slin24 to slin192, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 32000khz -> 8000khz' from format slin32 to slin, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 32000khz -> 12000khz' from format slin32 to slin12, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 32000khz -> 16000khz' from format slin32 to slin16, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 32000khz -> 24000khz' from format slin32 to slin24, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 32000khz -> 44100khz' from format slin32 to slin44, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 32000khz -> 48000khz' from format slin32 to slin48, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 32000khz -> 96000khz' from format slin32 to slin96, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 32000khz -> 192000khz' from format slin32 to slin192, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 44100khz -> 8000khz' from format slin44 to slin, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 44100khz -> 12000khz' from format slin44 to slin12, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 44100khz -> 16000khz' from format slin44 to slin16, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 44100khz -> 24000khz' from format slin44 to slin24, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 44100khz -> 32000khz' from format slin44 to slin32, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 44100khz -> 48000khz' from format slin44 to slin48, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 44100khz -> 96000khz' from format slin44 to slin96, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 44100khz -> 192000khz' from format slin44 to slin192, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 48000khz -> 8000khz' from format slin48 to slin, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 48000khz -> 12000khz' from format slin48 to slin12, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 48000khz -> 16000khz' from format slin48 to slin16, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 48000khz -> 24000khz' from format slin48 to slin24, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 48000khz -> 32000khz' from format slin48 to slin32, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 48000khz -> 44100khz' from format slin48 to slin44, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 48000khz -> 96000khz' from format slin48 to slin96, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 48000khz -> 192000khz' from format slin48 to slin192, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 96000khz -> 8000khz' from format slin96 to slin, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 96000khz -> 12000khz' from format slin96 to slin12, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 96000khz -> 16000khz' from format slin96 to slin16, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 96000khz -> 24000khz' from format slin96 to slin24, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 96000khz -> 32000khz' from format slin96 to slin32, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 96000khz -> 44100khz' from format slin96 to slin44, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 96000khz -> 48000khz' from format slin96 to slin48, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 96000khz -> 192000khz' from format slin96 to slin192, table cost, 800000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 192000khz -> 8000khz' from format slin192 to slin, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 192000khz -> 12000khz' from format slin192 to slin12, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 192000khz -> 16000khz' from format slin192 to slin16, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 192000khz -> 24000khz' from format slin192 to slin24, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 192000khz -> 32000khz' from format slin192 to slin32, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 192000khz -> 44100khz' from format slin192 to slin44, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 192000khz -> 48000khz' from format slin192 to slin48, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] translate.c: == Registered translator 'slin 192000khz -> 96000khz' from format slin192 to slin96, table cost, 850000, computational cost 999999 [Oct 10 13:52:23] VERBOSE[15559] loader.c: codec_resample.so => (SLIN Resampling Codec) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_celgenuserevent.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'CELGenUserEvent' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_celgenuserevent.so => (Generate an User-Defined CEL event) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading pbx_loopback.so. [Oct 10 13:52:23] VERBOSE[15559] loader.c: pbx_loopback.so => (Loopback Switch) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_flash.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Flash' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_flash.so => (Flash channel application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_exec.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Exec' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'TryExec' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'ExecIf' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_exec.so => (Executes dialplan applications) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading pbx_ael.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'AELSub' [Oct 10 13:52:23] NOTICE[15559] pbx_ael.c: Starting AEL load process. [Oct 10 13:52:23] NOTICE[15559] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Oct 10 13:52:23] NOTICE[15559] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Setting global variable 'CONSOLE-AEL' to '"Console/dsp"' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Setting global variable 'IAXINFO-AEL' to 'guest' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Setting global variable 'OUTBOUND-TRUNK' to '"Zap/g2"' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Setting global variable 'OUTBOUND-TRUNKMSD' to '1' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Registered extension context 'ael-builtin-h-bubble'; registrar: pbx_ael [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 1 to ael-builtin-h-bubble [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 9991 to ael-builtin-h-bubble [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 9992 to ael-builtin-h-bubble [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 9993 to ael-builtin-h-bubble [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 9994 to ael-builtin-h-bubble [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 9995 to ael-builtin-h-bubble [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 9996 to ael-builtin-h-bubble [Oct 10 13:52:23] NOTICE[15559] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Registered extension context 'parkedcalls'; registrar: res_parking [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_ael [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '700' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '701' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '702' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '703' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '704' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '705' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '706' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '707' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '708' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '709' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '710' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '711' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '712' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '713' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '714' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '715' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '716' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '717' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '718' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '719' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '720' priority 1 to parkedcalls [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Registered extension context 'internal'; registrar: pbx_config [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- merging incls/swits/igpats from old(internal) to new(internal) context, registrar = pbx_ael [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '201' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '201' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '201' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'h' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'phone_B' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'phone_A' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '666' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '999' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '999' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '999' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 7 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 6 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 5 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 4 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '998' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '997' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '997' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension 'agent-1001' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '555' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '555' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '553' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '552' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '552' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '551' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1003' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1003' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1003' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1003' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1002' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1002' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1002' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1002' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1001' priority 3 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1001' priority 2 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1001' priority 1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Added extension '1001' priority -1 to internal [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Registered extension context 'public'; registrar: pbx_config [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- merging incls/swits/igpats from old(public) to new(public) context, registrar = pbx_ael [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Registered extension context 'default'; registrar: pbx_config [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- merging incls/swits/igpats from old(default) to new(default) context, registrar = pbx_ael [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.000855 sec [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Time to restore hints and swap in new dialplan: 0.000009 sec [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Time to delete the old dialplan: 0.000092 sec [Oct 10 13:52:23] VERBOSE[15559] pbx.c: -- Total time merge_contexts_delete: 0.000956 sec [Oct 10 13:52:23] NOTICE[15559] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Oct 10 13:52:23] NOTICE[15559] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Oct 10 13:52:23] VERBOSE[15559] loader.c: pbx_ael.so => (Asterisk Extension Language Compiler) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_channel.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CHANNEL' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'CHANNELS' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'MASTER_CHANNEL' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_channel.so => (Channel information dialplan functions) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading func_lock.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'LOCK' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'TRYLOCK' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'UNLOCK' [Oct 10 13:52:23] VERBOSE[15559] loader.c: func_lock.so => (Dialplan mutexes) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading res_fax_spandsp.so. [Oct 10 13:52:23] VERBOSE[15559] res_fax.c: -- Registered handler for 'Spandsp' (Spandsp FAX Driver) [Oct 10 13:52:23] VERBOSE[15559] loader.c: res_fax_spandsp.so => (Spandsp G.711 and T.38 FAX Technologies) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_disa.so. [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'DISA' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_disa.so => (DISA (Direct Inward System Access) Application) [Oct 10 13:52:23] VERBOSE[15559] loader.c: Loading app_queue.so. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/queuerules.conf': Found [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/queues.conf': Found [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'Queue' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'AddQueueMember' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'RemoveQueueMember' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'PauseQueueMember' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'UnpauseQueueMember' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered application 'QueueLog' [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action Queues [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueueStatus [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueueSummary [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueueAdd [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueueRemove [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueuePause [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueueLog [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueuePenalty [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueueMemberRingInUse [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueueRule [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueueReload [Oct 10 13:52:23] VERBOSE[15559] manager.c: == Manager registered action QueueReset [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'QUEUE_VARIABLES' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'QUEUE_EXISTS' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'QUEUE_MEMBER' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'QUEUE_MEMBER_COUNT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'QUEUE_MEMBER_LIST' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'QUEUE_WAITING_COUNT' [Oct 10 13:52:23] VERBOSE[15559] pbx.c: == Registered custom function 'QUEUE_MEMBER_PENALTY' [Oct 10 13:52:23] VERBOSE[15559] loader.c: app_queue.so => (True Call Queueing) [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cli_permissions.conf': Found [Oct 10 13:52:23] VERBOSE[15559] asterisk.c: Asterisk Ready. [Oct 10 13:52:23] VERBOSE[15559] config.c: == Parsing '/etc/asterisk/cli.conf': Found [Oct 10 13:52:23] VERBOSE[15561] asterisk.c: -- Remote UNIX connection [Oct 10 13:52:29] Asterisk SVN-branch-12-r400824 built by root @ on a i686 running Linux on 2013-10-01 16:12:22 UTC [Oct 10 13:52:29] DEBUG[15708] config.c: Parsing /etc/asterisk/logger.conf [Oct 10 13:52:29] VERBOSE[15708] config.c: == Parsing '/etc/asterisk/logger.conf': Found [Oct 10 13:52:29] VERBOSE[15708] logger.c: Asterisk Queue Logger restarted [Oct 10 13:52:40] DEBUG[15619] logger.c: CALL_ID [C-00000000] created by thread. [Oct 10 13:52:40] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:52:40] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:52:40] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for y4064HUTe9QwgOPD6DQBKB0SFJeG7PXo - INVITE (No RTP) [Oct 10 13:52:40] DEBUG[15619][C-00000000] logger.c: CALL_ID [C-00000000] bound to thread. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 10 13:52:40] DEBUG[15619][C-00000000] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Oct 10 13:52:40] DEBUG[15619][C-00000000] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 10 13:52:40] DEBUG[15619][C-00000000] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 10 13:52:40] DEBUG[15619][C-00000000] sip/reqresp_parser.c: Found SIP option: -100rel- [Oct 10 13:52:40] DEBUG[15619][C-00000000] sip/reqresp_parser.c: Matched SIP option: 100rel [Oct 10 13:52:40] DEBUG[15619][C-00000000] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 10 13:52:40] DEBUG[15619][C-00000000] sip/reqresp_parser.c: Matched SIP option: timer [Oct 10 13:52:40] DEBUG[15619][C-00000000] sip/reqresp_parser.c: Found SIP option: -norefersub- [Oct 10 13:52:40] DEBUG[15619][C-00000000] sip/reqresp_parser.c: Matched SIP option: norefersub [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb2801f14' [Oct 10 13:52:40] DEBUG[15619][C-00000000] res_rtp_asterisk.c: Allocated port 13086 for RTP instance '0xb2801f14' [Oct 10 13:52:40] DEBUG[15619][C-00000000] pjsip: icess0xb2808dc ICE session created, comp_cnt=2, role is Unknown agent [Oct 10 13:52:40] DEBUG[15619][C-00000000] pjsip: icess0xb2808dc Candidate 0 added: comp_id=1, type=host, foundation=Ha1812a1, addr=10.24.18.161:13086, base=10.24.18.161:13086, prio=0x7effffff (2130706431) [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: RTP instance '0xb2801f14' is setup and ready to go [Oct 10 13:52:40] DEBUG[15619][C-00000000] pjsip: icess0xb2808dc Destroying ICE session 0xb2808dcc [Oct 10 13:52:40] DEBUG[15619][C-00000000] pjsip: ice_session.c ICE session 0xb2808dcc destroyed [Oct 10 13:52:40] DEBUG[15619][C-00000000] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb2801f14' [Oct 10 13:52:40] VERBOSE[15619][C-00000000] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Setting NAT on RTP to Off [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing session-level SDP o=- 76802451 76802451 IN IP4 10.24.19.97... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.19.97... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Setting payload 111 based on m type on 0xb2d91c78 [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Setting payload 18 based on m type on 0xb2d91c78 [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Setting payload 0 based on m type on 0xb2d91c78 [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Setting payload 58 based on m type on 0xb2d91c78 [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Setting payload 118 based on m type on 0xb2d91c78 [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Setting payload 8 based on m type on 0xb2d91c78 [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Setting payload 9 based on m type on 0xb2d91c78 [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Setting payload 58 based on m type on 0xb2d91c78 [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Setting payload 96 based on m type on 0xb2d91c78 [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4029 IN IP4 10.24.19.97... UNSUPPORTED OR FAILED. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 G726-32/8000... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:58 L16/16000... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:118 L16/8000... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:58 L16-256/16000... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Oct 10 13:52:40] DEBUG[15619][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb2801f14' [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Copying payload 0 from 0xb2d91c78 to 0xb280203c [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Copying payload 8 from 0xb2d91c78 to 0xb280203c [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Copying payload 9 from 0xb2d91c78 to 0xb280203c [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Copying payload 18 from 0xb2d91c78 to 0xb280203c [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Copying payload 58 from 0xb2d91c78 to 0xb280203c [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Copying payload 96 from 0xb2d91c78 to 0xb280203c [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Copying payload 111 from 0xb2d91c78 to 0xb280203c [Oct 10 13:52:40] DEBUG[15619][C-00000000] rtp_engine.c: Copying payload 118 from 0xb2d91c78 to 0xb280203c [Oct 10 13:52:40] DEBUG[15619][C-00000000] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb2801f14' [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: We're settling with these formats: (ulaw) [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Checking SIP call limits for device phone_A [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Updating call counter for incoming call [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: *** Our native formats are (ulaw) [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: *** Joint capabilities are (ulaw) [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: *** Our capabilities are (ulaw) [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: This channel will not be able to handle video. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Incoming INVITE with 'timer' option supported [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: INVITE also has "Session-Expires" header. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Session-Expires: 1800 [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: INVITE also has "Min-SE" header. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Received Min-SE: 90 [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Session timer started: 7 - y4064HUTe9QwgOPD6DQBKB0SFJeG7PXo 900000ms [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: SIP/phone_A-00000000: New call is still down.... Trying... [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:52:40] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 13:52:40] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 13:52:40] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 13:52:40] DEBUG[15619][C-00000000] logger.c: CALL_ID [C-00000000] being removed from thread. [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has changed to 'Not in use' [Oct 10 13:52:40] DEBUG[15709][C-00000000] logger.c: CALL_ID [C-00000000] bound to thread. [Oct 10 13:52:40] DEBUG[15681] app_queue.c: Device 'SIP/phone_A' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 10 13:52:40] DEBUG[15709][C-00000000] pbx.c: Launching 'NoOp' [Oct 10 13:52:40] VERBOSE[15709][C-00000000] pbx.c: -- Executing [201@internal:1] NoOp("SIP/phone_A-00000000", "") in new stack [Oct 10 13:52:40] DEBUG[15709][C-00000000] pbx.c: Launching 'Answer' [Oct 10 13:52:40] VERBOSE[15709][C-00000000] pbx.c: -- Executing [201@internal:2] Answer("SIP/phone_A-00000000", "") in new stack [Oct 10 13:52:40] DEBUG[15709][C-00000000] chan_sip.c: SIP answering channel: SIP/phone_A-00000000 [Oct 10 13:52:40] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 13:52:40] DEBUG[15709][C-00000000] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 10 13:52:40] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 13:52:40] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 13:52:40] DEBUG[15709][C-00000000] chan_sip.c: Setting framing from config on incoming call [Oct 10 13:52:40] DEBUG[15709][C-00000000] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Oct 10 13:52:40] DEBUG[15709][C-00000000] chan_sip.c: ** Our prefcodec: (nothing) [Oct 10 13:52:40] DEBUG[15709][C-00000000] chan_sip.c: -- Done with adding codecs to SDP [Oct 10 13:52:40] DEBUG[15709][C-00000000] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Oct 10 13:52:40] DEBUG[15709][C-00000000] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:52:40] DEBUG[15709][C-00000000] res_rtp_asterisk.c: 0xb2806430 -- Probation learning mode pass with source address 10.24.19.97:4028 [Oct 10 13:52:40] DEBUG[15709][C-00000000] pbx.c: Launching 'Stasis' [Oct 10 13:52:40] VERBOSE[15709][C-00000000] pbx.c: -- Executing [201@internal:3] Stasis("SIP/phone_A-00000000", "Bridge") in new stack [Oct 10 13:52:40] ERROR[15709][C-00000000] res_stasis.c: Stasis app 'Bridge' not registered [Oct 10 13:52:40] DEBUG[15709][C-00000000] pbx.c: Spawn extension (internal,201,3) exited non-zero on 'SIP/phone_A-00000000' [Oct 10 13:52:40] VERBOSE[15709][C-00000000] pbx.c: == Spawn extension (internal, 201, 3) exited non-zero on 'SIP/phone_A-00000000' [Oct 10 13:52:40] DEBUG[15709][C-00000000] channel.c: Soft-Hanging up channel 'SIP/phone_A-00000000' [Oct 10 13:52:40] DEBUG[15709][C-00000000] channel.c: Soft-Hanging up channel 'SIP/phone_A-00000000' [Oct 10 13:52:40] DEBUG[15709][C-00000000] pbx.c: Launching 'NoOp' [Oct 10 13:52:40] VERBOSE[15709][C-00000000] pbx.c: -- Executing [h@internal:1] NoOp("SIP/phone_A-00000000", "=== Hang up handler ===") in new stack [Oct 10 13:52:40] DEBUG[15709][C-00000000] pbx.c: Launching 'NoOp' [Oct 10 13:52:40] VERBOSE[15709][C-00000000] pbx.c: -- Executing [h@internal:2] NoOp("SIP/phone_A-00000000", "=== AGENT STATUS: ===") in new stack [Oct 10 13:52:40] DEBUG[15709][C-00000000] channel.c: Hanging up channel 'SIP/phone_A-00000000' [Oct 10 13:52:40] DEBUG[15709][C-00000000] chan_sip.c: Hangup call SIP/phone_A-00000000, SIP callid y4064HUTe9QwgOPD6DQBKB0SFJeG7PXo [Oct 10 13:52:40] DEBUG[15709][C-00000000] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb2801f14' [Oct 10 13:52:40] DEBUG[15709][C-00000000] chan_sip.c: Session timer stopped: 7 - y4064HUTe9QwgOPD6DQBKB0SFJeG7PXo [Oct 10 13:52:40] DEBUG[15568] cdr.c: Finalized CDR for SIP/phone_A-00000000 - start 1381431160.240726 answer 1381431160.241751 end 1381431160.527507 dispo ANSWERED [Oct 10 13:52:40] DEBUG[15568] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! [Oct 10 13:52:40] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 13:52:40] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 13:52:40] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 13:52:40] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 13:52:40] DEBUG[15568] res_config_sqlite.c: About to query table structure: SELECT sql FROM sqlite_master WHERE type='table' AND tbl_name='ast_cdr' [Oct 10 13:52:40] DEBUG[15568] res_config_sqlite.c: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"Phone A" <1001>','1001','h','internal','SIP/phone_A-00000000','NoOp','=== AGENT STATUS: ===','2013-10-10 13:52:40','2013-10-10 13:52:40','2013-10-10 13:52:40','0','0','ANSWERED','DOCUMENTATION','1381431160.0') [Oct 10 13:52:40] DEBUG[15619][C-00000000] logger.c: CALL_ID [C-00000000] bound to thread. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Stopping retransmission on 'y4064HUTe9QwgOPD6DQBKB0SFJeG7PXo' of Response 2745: Match Found [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Trying to put 'BYE sip:pho' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:52:40] DEBUG[15619][C-00000000] logger.c: CALL_ID [C-00000000] being removed from thread. [Oct 10 13:52:40] DEBUG[15619][C-00000000] logger.c: CALL_ID [C-00000000] bound to thread. [Oct 10 13:52:40] DEBUG[15619][C-00000000] chan_sip.c: Stopping retransmission on 'y4064HUTe9QwgOPD6DQBKB0SFJeG7PXo' of Request 102: Match Found [Oct 10 13:52:40] DEBUG[15619][C-00000000] logger.c: CALL_ID [C-00000000] being removed from thread. [Oct 10 13:52:40] DEBUG[15619] chan_sip.c: Destroying SIP dialog y4064HUTe9QwgOPD6DQBKB0SFJeG7PXo [Oct 10 13:52:40] DEBUG[15619] rtp_engine.c: Destroyed RTP instance '0xb2801f14' [Oct 10 13:52:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:52:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 13:52:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 13:52:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 13:52:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 13:52:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 13:52:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 13:52:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 13:52:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for SzM.QXCALzVQnFUlvKXkXk-VD3pWEcN9 - SUBSCRIBE (No RTP) [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:52:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 13:52:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog SzM.QXCALzVQnFUlvKXkXk-VD3pWEcN9 [Oct 10 13:53:06] DEBUG[15721] http.c: HTTP Request URI is /ari/events?app=Bridge [Oct 10 13:53:06] DEBUG[15721] http.c: match request [ari/events] with handler [httpstatus] len 10 [Oct 10 13:53:06] DEBUG[15721] http.c: match request [ari/events] with handler [phoneprov] len 9 [Oct 10 13:53:06] DEBUG[15721] http.c: match request [ari/events] with handler [static] len 6 [Oct 10 13:53:06] DEBUG[15721] http.c: match request [ari/events] with handler [ari] len 3 [Oct 10 13:53:06] DEBUG[15721] http.c: Match made with [ari] [Oct 10 13:53:14] DEBUG[15619] acl.c: For destination '10.24.17.17', our source address is '10.24.18.161'. [Oct 10 13:53:14] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:53:14] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 5256f79b557e-q8enuxwifjh3 - SUBSCRIBE (No RTP) [Oct 10 13:53:14] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:53:14] DEBUG[15619] chan_sip.c: build_route: Contact hop: ;reg-id=1 [Oct 10 13:53:14] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.17.17:5060 [Oct 10 13:53:14] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.17.17:5060 [Oct 10 13:53:14] DEBUG[15619] chan_sip.c: Acked pending invite 102 [Oct 10 13:53:14] DEBUG[15619] chan_sip.c: Stopping retransmission on '5256f79b557e-q8enuxwifjh3' of Request 102: Match Found [Oct 10 13:53:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:53:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 13:53:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 13:53:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 13:53:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 13:53:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 13:53:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 13:53:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has changed to 'Not in use' [Oct 10 13:53:17] DEBUG[15681] app_queue.c: Device 'SIP/phone_B' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 10 13:53:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for kIxrbYO1Ui0gdm2MeWgWsXNBFdVpHWnJ - SUBSCRIBE (No RTP) [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:53:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 13:53:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog kIxrbYO1Ui0gdm2MeWgWsXNBFdVpHWnJ [Oct 10 13:53:22] DEBUG[15582] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 10 13:53:22] DEBUG[15583] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 10 13:53:22] DEBUG[15579] threadpool.c: Destroying worker thread 1 [Oct 10 13:53:22] DEBUG[15579] threadpool.c: Destroying worker thread 2 [Oct 10 13:53:22] DEBUG[15584] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 10 13:53:22] DEBUG[15585] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 10 13:53:22] DEBUG[15579] threadpool.c: Destroying worker thread 3 [Oct 10 13:53:22] DEBUG[15579] threadpool.c: Destroying worker thread 4 [Oct 10 13:53:22] DEBUG[15586] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 10 13:53:22] DEBUG[15560] threadpool.c: Destroying worker thread 5 [Oct 10 13:53:22] DEBUG[15581] threadpool.c: Worker thread idle timeout reached. Dying. [Oct 10 13:53:22] DEBUG[15579] threadpool.c: Destroying worker thread 0 [Oct 10 13:53:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 13:53:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 13:53:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 13:53:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 13:53:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 13:53:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:53:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for vGBycL4hYDdAIhFROHjwDa1SXUBivF4x - SUBSCRIBE (No RTP) [Oct 10 13:53:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:53:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 13:53:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:53:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 13:53:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog vGBycL4hYDdAIhFROHjwDa1SXUBivF4x [Oct 10 13:54:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:54:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:54:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for bJQuixb50sj1fui-OLpgvOO13SZZeiWL - SUBSCRIBE (No RTP) [Oct 10 13:54:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:54:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 13:54:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:54:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 13:54:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog bJQuixb50sj1fui-OLpgvOO13SZZeiWL [Oct 10 13:55:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:55:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 13:55:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 13:55:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 13:55:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 13:55:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 13:55:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 13:55:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 13:55:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for guNUBWbJHJ7Jtyzz6UaX3CFcVA1pEaCJ - SUBSCRIBE (No RTP) [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:55:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 13:55:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog guNUBWbJHJ7Jtyzz6UaX3CFcVA1pEaCJ [Oct 10 13:55:46] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:55:46] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 13:55:46] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 13:55:46] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 13:55:46] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 13:55:46] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 13:55:46] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 13:55:46] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 13:55:46] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for TrZU4FKZyVLIdtZ07ltBWLRoAp.Pa2-E - SUBSCRIBE (No RTP) [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:55:46] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 13:55:46] DEBUG[15619] chan_sip.c: Destroying SIP dialog TrZU4FKZyVLIdtZ07ltBWLRoAp.Pa2-E [Oct 10 13:55:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 13:55:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 13:56:18] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 13:56:18] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 13:57:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:57:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 13:57:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 13:57:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 13:57:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 13:57:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 13:57:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 13:57:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 13:57:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for KG1KNBlZ65hhMjxm7N-7Dr5XqFDKzets - SUBSCRIBE (No RTP) [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:57:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 13:57:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog KG1KNBlZ65hhMjxm7N-7Dr5XqFDKzets [Oct 10 13:58:16] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:58:16] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 13:58:16] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 13:58:16] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 13:58:16] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 13:58:16] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 13:58:16] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 13:58:16] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 13:58:16] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for -HJm9R5PTva.epuo2xNi7itZxo1xdE7G - SUBSCRIBE (No RTP) [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:58:16] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 13:58:16] DEBUG[15619] chan_sip.c: Destroying SIP dialog -HJm9R5PTva.epuo2xNi7itZxo1xdE7G [Oct 10 13:58:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 13:58:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 13:58:48] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 13:58:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 13:58:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 13:58:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:58:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for VWKfinDfnKTTSAFIvioNlxLKsnuZ-8QC - SUBSCRIBE (No RTP) [Oct 10 13:58:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:58:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 13:58:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:58:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 13:58:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog VWKfinDfnKTTSAFIvioNlxLKsnuZ-8QC [Oct 10 13:59:31] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:59:31] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:59:31] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for .QLi7PHdhKVR-afZqYzyC7y3vgM6Xl3V - SUBSCRIBE (No RTP) [Oct 10 13:59:31] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:59:31] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:59:32] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:59:32] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:59:32] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for PLockzot5wXYYBQ89eWd18qPrTFlxMPl - SUBSCRIBE (No RTP) [Oct 10 13:59:32] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:59:32] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 13:59:32] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 13:59:32] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:59:32] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 13:59:32] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:59:32] DEBUG[15619] chan_sip.c: Acked pending invite 102 [Oct 10 13:59:32] DEBUG[15619] chan_sip.c: Stopping retransmission on 'PLockzot5wXYYBQ89eWd18qPrTFlxMPl' of Request 102: Match Found [Oct 10 13:59:47] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 13:59:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:59:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for X2jdJCTY-rOhjkQ6hSkazZmb5.DFWr4Y - SUBSCRIBE (No RTP) [Oct 10 13:59:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:59:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 13:59:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 13:59:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 13:59:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog X2jdJCTY-rOhjkQ6hSkazZmb5.DFWr4Y [Oct 10 13:59:53] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 13:59:53] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:59:53] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QoUDni7AxIVFvNW7RKIijSZKly11NsAD - SUBSCRIBE (No RTP) [Oct 10 13:59:53] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:59:53] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:59:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 13:59:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 13:59:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for EVj1DqKxYCA3vR-L.ceVXGwe5iq0.Ixi - SUBSCRIBE (No RTP) [Oct 10 13:59:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 13:59:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 13:59:54] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 13:59:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:59:54] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 13:59:54] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 13:59:54] DEBUG[15619] chan_sip.c: Acked pending invite 102 [Oct 10 13:59:54] DEBUG[15619] chan_sip.c: Stopping retransmission on 'EVj1DqKxYCA3vR-L.ceVXGwe5iq0.Ixi' of Request 102: Match Found [Oct 10 14:00:03] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog '.QLi7PHdhKVR-afZqYzyC7y3vgM6Xl3V' [Oct 10 14:00:03] DEBUG[15619] chan_sip.c: Destroying SIP dialog .QLi7PHdhKVR-afZqYzyC7y3vgM6Xl3V [Oct 10 14:00:05] DEBUG[15795] http.c: HTTP Request URI is /ari/events?app=Bridge [Oct 10 14:00:05] DEBUG[15795] http.c: match request [ari/events] with handler [httpstatus] len 10 [Oct 10 14:00:05] DEBUG[15795] http.c: match request [ari/events] with handler [phoneprov] len 9 [Oct 10 14:00:05] DEBUG[15795] http.c: match request [ari/events] with handler [static] len 6 [Oct 10 14:00:05] DEBUG[15795] http.c: match request [ari/events] with handler [ari] len 3 [Oct 10 14:00:05] DEBUG[15795] http.c: Match made with [ari] [Oct 10 14:00:12] DEBUG[15803] http.c: HTTP Request URI is /ari/events?app=Bridge [Oct 10 14:00:12] DEBUG[15803] http.c: match request [ari/events] with handler [httpstatus] len 10 [Oct 10 14:00:12] DEBUG[15803] http.c: match request [ari/events] with handler [phoneprov] len 9 [Oct 10 14:00:12] DEBUG[15803] http.c: match request [ari/events] with handler [static] len 6 [Oct 10 14:00:12] DEBUG[15803] http.c: match request [ari/events] with handler [ari] len 3 [Oct 10 14:00:12] DEBUG[15803] http.c: Match made with [ari] [Oct 10 14:00:25] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QoUDni7AxIVFvNW7RKIijSZKly11NsAD' [Oct 10 14:00:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog QoUDni7AxIVFvNW7RKIijSZKly11NsAD [Oct 10 14:00:26] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:00:26] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:00:26] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:00:26] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:00:26] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:00:26] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:00:26] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:00:26] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:00:26] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for tcKPRxKfZ0j1pt4FFX2C7Xx7M0yJMdkt - SUBSCRIBE (No RTP) [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:00:26] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:00:26] DEBUG[15619] chan_sip.c: Destroying SIP dialog tcKPRxKfZ0j1pt4FFX2C7Xx7M0yJMdkt [Oct 10 14:00:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:00:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:00:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:00:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:00:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:00:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:00:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:00:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:00:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for B2K7tQ6DjLNyIXTMOBcBztLAg8dcTJ4l - SUBSCRIBE (No RTP) [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:00:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:00:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog B2K7tQ6DjLNyIXTMOBcBztLAg8dcTJ4l [Oct 10 14:00:58] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:00:58] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:01:13] DEBUG[15816] http.c: HTTP Request URI is /ari/events?app=Bridge [Oct 10 14:01:13] DEBUG[15816] http.c: match request [ari/events] with handler [httpstatus] len 10 [Oct 10 14:01:13] DEBUG[15816] http.c: match request [ari/events] with handler [phoneprov] len 9 [Oct 10 14:01:13] DEBUG[15816] http.c: match request [ari/events] with handler [static] len 6 [Oct 10 14:01:13] DEBUG[15816] http.c: match request [ari/events] with handler [ari] len 3 [Oct 10 14:01:13] DEBUG[15816] http.c: Match made with [ari] [Oct 10 14:01:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:01:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:01:21] DEBUG[15823] http.c: HTTP Request URI is / [Oct 10 14:01:21] DEBUG[15823] http.c: match request [] with handler [httpstatus] len 10 [Oct 10 14:01:21] DEBUG[15823] http.c: match request [] with handler [phoneprov] len 9 [Oct 10 14:01:21] DEBUG[15823] http.c: match request [] with handler [static] len 6 [Oct 10 14:01:21] DEBUG[15823] http.c: match request [] with handler [ari] len 3 [Oct 10 14:01:21] DEBUG[15823] http.c: match request [] with handler [ws] len 2 [Oct 10 14:01:21] DEBUG[15823] http.c: Requested URI [] has no handler [Oct 10 14:01:26] DEBUG[15830] http.c: HTTP Request URI is / [Oct 10 14:01:26] DEBUG[15830] http.c: match request [] with handler [httpstatus] len 10 [Oct 10 14:01:26] DEBUG[15830] http.c: match request [] with handler [phoneprov] len 9 [Oct 10 14:01:26] DEBUG[15830] http.c: match request [] with handler [static] len 6 [Oct 10 14:01:26] DEBUG[15830] http.c: match request [] with handler [ari] len 3 [Oct 10 14:01:26] DEBUG[15830] http.c: match request [] with handler [ws] len 2 [Oct 10 14:01:26] DEBUG[15830] http.c: Requested URI [] has no handler [Oct 10 14:01:40] DEBUG[15833] http.c: HTTP Request URI is / [Oct 10 14:01:40] DEBUG[15833] http.c: match request [] with handler [httpstatus] len 10 [Oct 10 14:01:40] DEBUG[15833] http.c: match request [] with handler [phoneprov] len 9 [Oct 10 14:01:40] DEBUG[15833] http.c: match request [] with handler [static] len 6 [Oct 10 14:01:40] DEBUG[15833] http.c: match request [] with handler [ari] len 3 [Oct 10 14:01:40] DEBUG[15833] http.c: match request [] with handler [ws] len 2 [Oct 10 14:01:40] DEBUG[15833] http.c: Requested URI [] has no handler [Oct 10 14:01:53] DEBUG[15838] http.c: HTTP Request URI is / [Oct 10 14:01:53] DEBUG[15838] http.c: match request [] with handler [httpstatus] len 10 [Oct 10 14:01:53] DEBUG[15838] http.c: match request [] with handler [phoneprov] len 9 [Oct 10 14:01:53] DEBUG[15838] http.c: match request [] with handler [static] len 6 [Oct 10 14:01:53] DEBUG[15838] http.c: match request [] with handler [ari] len 3 [Oct 10 14:01:53] DEBUG[15838] http.c: match request [] with handler [ws] len 2 [Oct 10 14:01:53] DEBUG[15838] http.c: Requested URI [] has no handler [Oct 10 14:01:56] DEBUG[15841] http.c: HTTP Request URI is / [Oct 10 14:01:56] DEBUG[15841] http.c: match request [] with handler [httpstatus] len 10 [Oct 10 14:01:56] DEBUG[15841] http.c: match request [] with handler [phoneprov] len 9 [Oct 10 14:01:56] DEBUG[15841] http.c: match request [] with handler [static] len 6 [Oct 10 14:01:56] DEBUG[15841] http.c: match request [] with handler [ari] len 3 [Oct 10 14:01:56] DEBUG[15841] http.c: match request [] with handler [ws] len 2 [Oct 10 14:01:56] DEBUG[15841] http.c: Requested URI [] has no handler [Oct 10 14:01:58] DEBUG[15845] http.c: HTTP Request URI is / [Oct 10 14:01:58] DEBUG[15845] http.c: match request [] with handler [httpstatus] len 10 [Oct 10 14:01:58] DEBUG[15845] http.c: match request [] with handler [phoneprov] len 9 [Oct 10 14:01:58] DEBUG[15845] http.c: match request [] with handler [static] len 6 [Oct 10 14:01:58] DEBUG[15845] http.c: match request [] with handler [ari] len 3 [Oct 10 14:01:58] DEBUG[15845] http.c: match request [] with handler [ws] len 2 [Oct 10 14:01:58] DEBUG[15845] http.c: Requested URI [] has no handler [Oct 10 14:02:25] DEBUG[15849] http.c: HTTP Request URI is /ari/events?app=Bridge [Oct 10 14:02:25] DEBUG[15849] http.c: match request [ari/events] with handler [httpstatus] len 10 [Oct 10 14:02:25] DEBUG[15849] http.c: match request [ari/events] with handler [phoneprov] len 9 [Oct 10 14:02:25] DEBUG[15849] http.c: match request [ari/events] with handler [static] len 6 [Oct 10 14:02:25] DEBUG[15849] http.c: match request [ari/events] with handler [ari] len 3 [Oct 10 14:02:25] DEBUG[15849] http.c: Match made with [ari] [Oct 10 14:02:34] DEBUG[15852] http.c: HTTP Request URI is /ari/events?app=Bridge [Oct 10 14:02:34] DEBUG[15852] http.c: match request [ari/events] with handler [httpstatus] len 10 [Oct 10 14:02:34] DEBUG[15852] http.c: match request [ari/events] with handler [phoneprov] len 9 [Oct 10 14:02:34] DEBUG[15852] http.c: match request [ari/events] with handler [static] len 6 [Oct 10 14:02:34] DEBUG[15852] http.c: match request [ari/events] with handler [ari] len 3 [Oct 10 14:02:34] DEBUG[15852] http.c: Match made with [ari] [Oct 10 14:02:41] DEBUG[15855] http.c: HTTP Request URI is /ari/events?app=Bridge [Oct 10 14:02:41] DEBUG[15855] http.c: match request [ari/events] with handler [httpstatus] len 10 [Oct 10 14:02:41] DEBUG[15855] http.c: match request [ari/events] with handler [phoneprov] len 9 [Oct 10 14:02:41] DEBUG[15855] http.c: match request [ari/events] with handler [static] len 6 [Oct 10 14:02:41] DEBUG[15855] http.c: match request [ari/events] with handler [ari] len 3 [Oct 10 14:02:41] DEBUG[15855] http.c: Match made with [ari] [Oct 10 14:02:56] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:02:56] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:02:56] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:02:56] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:02:56] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:02:56] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:02:56] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:02:56] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:02:56] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for anLEO82705E6hpAssAA2a-KPOA0H04yE - SUBSCRIBE (No RTP) [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:02:56] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:02:56] DEBUG[15619] chan_sip.c: Destroying SIP dialog anLEO82705E6hpAssAA2a-KPOA0H04yE [Oct 10 14:03:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:03:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:03:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:03:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:03:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:03:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:03:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:03:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:03:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for y7ifyDKZoOG8XGHeWAaN0f4zJonYodqt - SUBSCRIBE (No RTP) [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:03:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:03:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog y7ifyDKZoOG8XGHeWAaN0f4zJonYodqt [Oct 10 14:03:28] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:03:28] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:03:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:03:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:03:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:03:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:03:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 0VYBs2R0Pg97w6V5tF3tI4AItcCHqOMs - SUBSCRIBE (No RTP) [Oct 10 14:03:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:03:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:03:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:03:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:03:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog 0VYBs2R0Pg97w6V5tF3tI4AItcCHqOMs [Oct 10 14:04:06] DEBUG[15876] http.c: HTTP Request URI is /ari/events?app=Bridge&api_key=admin%3Asecret [Oct 10 14:04:06] DEBUG[15876] http.c: match request [ari/events] with handler [httpstatus] len 10 [Oct 10 14:04:06] DEBUG[15876] http.c: match request [ari/events] with handler [phoneprov] len 9 [Oct 10 14:04:06] DEBUG[15876] http.c: match request [ari/events] with handler [static] len 6 [Oct 10 14:04:06] DEBUG[15876] http.c: match request [ari/events] with handler [ari] len 3 [Oct 10 14:04:06] DEBUG[15876] http.c: Match made with [ari] [Oct 10 14:04:06] DEBUG[15876] res_ari.c: Finding handler for events [Oct 10 14:04:06] DEBUG[15876] res_ari.c: Checking endpoints [Oct 10 14:04:06] DEBUG[15876] res_ari.c: Checking channels [Oct 10 14:04:06] DEBUG[15876] res_ari.c: Checking events [Oct 10 14:04:06] DEBUG[15876] res_ari.c: Got it! [Oct 10 14:04:06] VERBOSE[15876] res_http_websocket.c: == WebSocket connection from '10.24.18.161:60078' for protocol '' accepted using version '13' [Oct 10 14:04:06] DEBUG[15876] ari/resource_events.c: /events WebSocket connection [Oct 10 14:04:06] VERBOSE[15876] stasis/app.c: Creating Stasis app 'Bridge' [Oct 10 14:04:23] WARNING[15876] ari/ari_websockets.c: WebSocket input failed to parse [Oct 10 14:04:32] ERROR[15876] /usr/src/asterisk/12/asterisk-12-branch/include/asterisk/utils.h: Memory Allocation Failure in function __ast_websocket_read at line 452 of res_http_websocket.c [Oct 10 14:04:32] WARNING[15876] ari/ari_websockets.c: WebSocket input failed to parse [Oct 10 14:04:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:04:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:04:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for fgVVgfey84fE035n9LjKNEBAaOoWtPKo - SUBSCRIBE (No RTP) [Oct 10 14:04:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:04:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:04:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:04:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:04:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog fgVVgfey84fE035n9LjKNEBAaOoWtPKo [Oct 10 14:05:22] DEBUG[15887] http.c: HTTP Request URI is /ari/events?app=Bridge&api_key=admin%3Asecret [Oct 10 14:05:22] DEBUG[15887] http.c: match request [ari/events] with handler [httpstatus] len 10 [Oct 10 14:05:22] DEBUG[15887] http.c: match request [ari/events] with handler [phoneprov] len 9 [Oct 10 14:05:22] DEBUG[15887] http.c: match request [ari/events] with handler [static] len 6 [Oct 10 14:05:22] DEBUG[15887] http.c: match request [ari/events] with handler [ari] len 3 [Oct 10 14:05:22] DEBUG[15887] http.c: Match made with [ari] [Oct 10 14:05:22] DEBUG[15887] res_ari.c: Finding handler for events [Oct 10 14:05:22] DEBUG[15887] res_ari.c: Checking endpoints [Oct 10 14:05:22] DEBUG[15887] res_ari.c: Checking channels [Oct 10 14:05:22] DEBUG[15887] res_ari.c: Checking events [Oct 10 14:05:22] DEBUG[15887] res_ari.c: Got it! [Oct 10 14:05:22] VERBOSE[15887] res_http_websocket.c: == WebSocket connection from '10.24.18.161:60079' for protocol '' accepted using version '13' [Oct 10 14:05:22] DEBUG[15887] ari/resource_events.c: /events WebSocket connection [Oct 10 14:05:22] VERBOSE[15887] stasis/app.c: Replacing Stasis app 'Bridge' [Oct 10 14:05:26] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:05:26] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:05:26] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:05:26] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:05:26] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:05:26] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:05:26] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:05:26] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:05:26] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for qEb0peB3C60u8cTZMWr4DwwDPNTGhcxU - SUBSCRIBE (No RTP) [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:05:26] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:05:26] DEBUG[15619] chan_sip.c: Destroying SIP dialog qEb0peB3C60u8cTZMWr4DwwDPNTGhcxU [Oct 10 14:05:43] DEBUG[15619] logger.c: CALL_ID [C-00000001] created by thread. [Oct 10 14:05:43] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:05:43] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:05:43] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX - INVITE (No RTP) [Oct 10 14:05:43] DEBUG[15619][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 10 14:05:43] DEBUG[15619][C-00000001] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Oct 10 14:05:43] DEBUG[15619][C-00000001] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 10 14:05:43] DEBUG[15619][C-00000001] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 10 14:05:43] DEBUG[15619][C-00000001] sip/reqresp_parser.c: Found SIP option: -100rel- [Oct 10 14:05:43] DEBUG[15619][C-00000001] sip/reqresp_parser.c: Matched SIP option: 100rel [Oct 10 14:05:43] DEBUG[15619][C-00000001] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 10 14:05:43] DEBUG[15619][C-00000001] sip/reqresp_parser.c: Matched SIP option: timer [Oct 10 14:05:43] DEBUG[15619][C-00000001] sip/reqresp_parser.c: Found SIP option: -norefersub- [Oct 10 14:05:43] DEBUG[15619][C-00000001] sip/reqresp_parser.c: Matched SIP option: norefersub [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb2a0a374' [Oct 10 14:05:43] DEBUG[15619][C-00000001] res_rtp_asterisk.c: Allocated port 15304 for RTP instance '0xb2a0a374' [Oct 10 14:05:43] DEBUG[15619][C-00000001] pjsip: icess0xb2a02ea ICE session created, comp_cnt=2, role is Unknown agent [Oct 10 14:05:43] DEBUG[15619][C-00000001] pjsip: icess0xb2a02ea Candidate 0 added: comp_id=1, type=host, foundation=Ha1812a1, addr=10.24.18.161:15304, base=10.24.18.161:15304, prio=0x7effffff (2130706431) [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: RTP instance '0xb2a0a374' is setup and ready to go [Oct 10 14:05:43] DEBUG[15619][C-00000001] pjsip: icess0xb2a02ea Destroying ICE session 0xb2a02eac [Oct 10 14:05:43] DEBUG[15619][C-00000001] pjsip: ice_session.c ICE session 0xb2a02eac destroyed [Oct 10 14:05:43] DEBUG[15619][C-00000001] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb2a0a374' [Oct 10 14:05:43] VERBOSE[15619][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Setting NAT on RTP to Off [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP o=- 76803233 76803233 IN IP4 10.24.19.97... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.19.97... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 111 based on m type on 0xb2d91c78 [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 18 based on m type on 0xb2d91c78 [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0xb2d91c78 [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 58 based on m type on 0xb2d91c78 [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 118 based on m type on 0xb2d91c78 [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 8 based on m type on 0xb2d91c78 [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 9 based on m type on 0xb2d91c78 [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 58 based on m type on 0xb2d91c78 [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0xb2d91c78 [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4031 IN IP4 10.24.19.97... UNSUPPORTED OR FAILED. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 G726-32/8000... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:58 L16/16000... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:118 L16/8000... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:58 L16-256/16000... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Oct 10 14:05:43] DEBUG[15619][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb2a0a374' [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 0 from 0xb2d91c78 to 0xb2a0a49c [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 8 from 0xb2d91c78 to 0xb2a0a49c [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 9 from 0xb2d91c78 to 0xb2a0a49c [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 18 from 0xb2d91c78 to 0xb2a0a49c [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 58 from 0xb2d91c78 to 0xb2a0a49c [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 96 from 0xb2d91c78 to 0xb2a0a49c [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 111 from 0xb2d91c78 to 0xb2a0a49c [Oct 10 14:05:43] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 118 from 0xb2d91c78 to 0xb2a0a49c [Oct 10 14:05:43] DEBUG[15619][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb2a0a374' [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Checking SIP call limits for device phone_A [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Updating call counter for incoming call [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: *** Our native formats are (ulaw) [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: *** Joint capabilities are (ulaw) [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: *** Our capabilities are (ulaw) [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: This channel will not be able to handle video. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Incoming INVITE with 'timer' option supported [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: INVITE also has "Session-Expires" header. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Session-Expires: 1800 [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: INVITE also has "Min-SE" header. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Received Min-SE: 90 [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Session timer started: 44 - iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX 900000ms [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: SIP/phone_A-00000001: New call is still down.... Trying... [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:05:43] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:05:43] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:05:43] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:05:43] DEBUG[15619][C-00000001] logger.c: CALL_ID [C-00000001] being removed from thread. [Oct 10 14:05:43] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:05:43] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:05:43] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:05:43] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:05:43] DEBUG[15888][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:05:43] DEBUG[15888][C-00000001] pbx.c: Launching 'NoOp' [Oct 10 14:05:43] VERBOSE[15888][C-00000001] pbx.c: -- Executing [201@internal:1] NoOp("SIP/phone_A-00000001", "") in new stack [Oct 10 14:05:43] DEBUG[15888][C-00000001] pbx.c: Launching 'Answer' [Oct 10 14:05:43] VERBOSE[15888][C-00000001] pbx.c: -- Executing [201@internal:2] Answer("SIP/phone_A-00000001", "") in new stack [Oct 10 14:05:43] DEBUG[15888][C-00000001] chan_sip.c: SIP answering channel: SIP/phone_A-00000001 [Oct 10 14:05:43] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:05:43] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 10 14:05:43] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:05:43] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:05:43] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:05:43] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:05:43] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:05:43] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:05:43] DEBUG[15888][C-00000001] chan_sip.c: Setting framing from config on incoming call [Oct 10 14:05:43] DEBUG[15888][C-00000001] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Oct 10 14:05:43] DEBUG[15888][C-00000001] chan_sip.c: ** Our prefcodec: (nothing) [Oct 10 14:05:43] DEBUG[15888][C-00000001] chan_sip.c: -- Done with adding codecs to SDP [Oct 10 14:05:43] DEBUG[15888][C-00000001] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Oct 10 14:05:43] DEBUG[15888][C-00000001] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:05:43] DEBUG[15888][C-00000001] res_rtp_asterisk.c: 0xb2cc9aa8 -- Probation learning mode pass with source address 10.24.19.97:4030 [Oct 10 14:05:43] DEBUG[15888][C-00000001] pbx.c: Launching 'Stasis' [Oct 10 14:05:43] VERBOSE[15888][C-00000001] pbx.c: -- Executing [201@internal:3] Stasis("SIP/phone_A-00000001", "Bridge") in new stack [Oct 10 14:05:43] DEBUG[15619][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 10 14:05:43] DEBUG[15619][C-00000001] chan_sip.c: Stopping retransmission on 'iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX' of Response 23567: Match Found [Oct 10 14:05:43] DEBUG[15619][C-00000001] logger.c: CALL_ID [C-00000001] being removed from thread. [Oct 10 14:05:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:05:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:05:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:05:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:05:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:05:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:05:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:05:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:05:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:05:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for xrbW13.NJ4Mm.SNYuAI5RnWwf9MYocyz - SUBSCRIBE (No RTP) [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:05:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:05:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog xrbW13.NJ4Mm.SNYuAI5RnWwf9MYocyz [Oct 10 14:05:51] DEBUG[15619] logger.c: CALL_ID [C-00000002] created by thread. [Oct 10 14:05:51] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:05:51] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:05:51] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ - INVITE (No RTP) [Oct 10 14:05:51] DEBUG[15619][C-00000002] logger.c: CALL_ID [C-00000002] bound to thread. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE [Oct 10 14:05:51] DEBUG[15619][C-00000002] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, 100rel, timer, norefersub" [Oct 10 14:05:51] DEBUG[15619][C-00000002] sip/reqresp_parser.c: Found SIP option: -replaces- [Oct 10 14:05:51] DEBUG[15619][C-00000002] sip/reqresp_parser.c: Matched SIP option: replaces [Oct 10 14:05:51] DEBUG[15619][C-00000002] sip/reqresp_parser.c: Found SIP option: -100rel- [Oct 10 14:05:51] DEBUG[15619][C-00000002] sip/reqresp_parser.c: Matched SIP option: 100rel [Oct 10 14:05:51] DEBUG[15619][C-00000002] sip/reqresp_parser.c: Found SIP option: -timer- [Oct 10 14:05:51] DEBUG[15619][C-00000002] sip/reqresp_parser.c: Matched SIP option: timer [Oct 10 14:05:51] DEBUG[15619][C-00000002] sip/reqresp_parser.c: Found SIP option: -norefersub- [Oct 10 14:05:51] DEBUG[15619][C-00000002] sip/reqresp_parser.c: Matched SIP option: norefersub [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x912892c' [Oct 10 14:05:51] DEBUG[15619][C-00000002] res_rtp_asterisk.c: Allocated port 14890 for RTP instance '0x912892c' [Oct 10 14:05:51] DEBUG[15619][C-00000002] pjsip: icess0x91d3adc ICE session created, comp_cnt=2, role is Unknown agent [Oct 10 14:05:51] DEBUG[15619][C-00000002] pjsip: icess0x91d3adc Candidate 0 added: comp_id=1, type=host, foundation=Ha1812a1, addr=10.24.18.161:14890, base=10.24.18.161:14890, prio=0x7effffff (2130706431) [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: RTP instance '0x912892c' is setup and ready to go [Oct 10 14:05:51] DEBUG[15619][C-00000002] pjsip: icess0x91d3adc Destroying ICE session 0x91d3adc [Oct 10 14:05:51] DEBUG[15619][C-00000002] pjsip: ice_session.c ICE session 0x91d3adc destroyed [Oct 10 14:05:51] DEBUG[15619][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x912892c' [Oct 10 14:05:51] VERBOSE[15619][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5 [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Setting NAT on RTP to Off [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP o=- 76803241 76803241 IN IP4 10.24.18.165... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.165... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0xb2d91c78 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 8 based on m type on 0xb2d91c78 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 9 based on m type on 0xb2d91c78 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 111 based on m type on 0xb2d91c78 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 18 based on m type on 0xb2d91c78 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 58 based on m type on 0xb2d91c78 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 118 based on m type on 0xb2d91c78 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 58 based on m type on 0xb2d91c78 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0xb2d91c78 [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4013 IN IP4 10.24.18.165... UNSUPPORTED OR FAILED. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 G726-32/8000... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:58 L16/16000... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:118 L16/8000... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:58 L16-256/16000... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Oct 10 14:05:51] DEBUG[15619][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x912892c' [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 0 from 0xb2d91c78 to 0x9128a54 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 8 from 0xb2d91c78 to 0x9128a54 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 9 from 0xb2d91c78 to 0x9128a54 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 18 from 0xb2d91c78 to 0x9128a54 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 58 from 0xb2d91c78 to 0x9128a54 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 96 from 0xb2d91c78 to 0x9128a54 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 111 from 0xb2d91c78 to 0x9128a54 [Oct 10 14:05:51] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 118 from 0xb2d91c78 to 0x9128a54 [Oct 10 14:05:51] DEBUG[15619][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x912892c' [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Checking SIP call limits for device phone_B [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Updating call counter for incoming call [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: *** Our native formats are (ulaw) [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: *** Joint capabilities are (ulaw) [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: *** Our capabilities are (ulaw) [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: This channel will not be able to handle video. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Incoming INVITE with 'timer' option supported [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: INVITE also has "Session-Expires" header. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Session-Expires: 1800 [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: INVITE also has "Min-SE" header. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Received Min-SE: 90 [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Session timer started: 50 - -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ 900000ms [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: SIP/phone_B-00000002: New call is still down.... Trying... [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:05:51] DEBUG[15619][C-00000002] logger.c: CALL_ID [C-00000002] being removed from thread. [Oct 10 14:05:51] DEBUG[15890][C-00000002] logger.c: CALL_ID [C-00000002] bound to thread. [Oct 10 14:05:51] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:05:51] DEBUG[15890][C-00000002] pbx.c: Launching 'NoOp' [Oct 10 14:05:51] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:05:51] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:05:51] VERBOSE[15890][C-00000002] pbx.c: -- Executing [201@internal:1] NoOp("SIP/phone_B-00000002", "") in new stack [Oct 10 14:05:51] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:05:51] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:05:51] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:05:51] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:05:51] DEBUG[15890][C-00000002] pbx.c: Launching 'Answer' [Oct 10 14:05:51] VERBOSE[15890][C-00000002] pbx.c: -- Executing [201@internal:2] Answer("SIP/phone_B-00000002", "") in new stack [Oct 10 14:05:51] DEBUG[15890][C-00000002] chan_sip.c: SIP answering channel: SIP/phone_B-00000002 [Oct 10 14:05:51] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:05:51] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 10 14:05:51] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:05:51] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:05:51] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:05:51] DEBUG[15890][C-00000002] chan_sip.c: Setting framing from config on incoming call [Oct 10 14:05:51] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:05:51] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:05:51] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:05:51] DEBUG[15890][C-00000002] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Oct 10 14:05:51] DEBUG[15890][C-00000002] chan_sip.c: ** Our prefcodec: (nothing) [Oct 10 14:05:51] DEBUG[15890][C-00000002] chan_sip.c: -- Done with adding codecs to SDP [Oct 10 14:05:51] DEBUG[15890][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Oct 10 14:05:51] DEBUG[15890][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:05:51] DEBUG[15890][C-00000002] res_rtp_asterisk.c: 0x912dec0 -- Probation learning mode pass with source address 10.24.18.165:4012 [Oct 10 14:05:51] DEBUG[15890][C-00000002] pbx.c: Launching 'Stasis' [Oct 10 14:05:51] VERBOSE[15890][C-00000002] pbx.c: -- Executing [201@internal:3] Stasis("SIP/phone_B-00000002", "Bridge") in new stack [Oct 10 14:05:51] DEBUG[15619][C-00000002] logger.c: CALL_ID [C-00000002] bound to thread. [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: **** Received ACK (6) - Command in SIP ACK [Oct 10 14:05:51] DEBUG[15619][C-00000002] chan_sip.c: Stopping retransmission on '-gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ' of Response 17170: Match Found [Oct 10 14:05:51] DEBUG[15619][C-00000002] logger.c: CALL_ID [C-00000002] being removed from thread. [Oct 10 14:05:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:05:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:05:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:05:58] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:05:58] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:05:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:18] DEBUG[15893] http.c: HTTP Request URI is /ari/api-docs/resources.json?api_key=admin:secret [Oct 10 14:06:18] DEBUG[15893] http.c: match request [ari/api-docs/resources.json] with handler [httpstatus] len 10 [Oct 10 14:06:18] DEBUG[15893] http.c: match request [ari/api-docs/resources.json] with handler [phoneprov] len 9 [Oct 10 14:06:18] DEBUG[15893] http.c: match request [ari/api-docs/resources.json] with handler [static] len 6 [Oct 10 14:06:18] DEBUG[15893] http.c: match request [ari/api-docs/resources.json] with handler [ari] len 3 [Oct 10 14:06:18] DEBUG[15893] http.c: Match made with [ari] [Oct 10 14:06:18] DEBUG[15893] res_ari.c: ast_ari_get_docs(resources.json) [Oct 10 14:06:18] DEBUG[15894] http.c: HTTP Request URI is /ari/api-docs/asterisk.json?api_key=admin:secret [Oct 10 14:06:18] DEBUG[15894] http.c: match request [ari/api-docs/asterisk.json] with handler [httpstatus] len 10 [Oct 10 14:06:18] DEBUG[15894] http.c: match request [ari/api-docs/asterisk.json] with handler [phoneprov] len 9 [Oct 10 14:06:18] DEBUG[15894] http.c: match request [ari/api-docs/asterisk.json] with handler [static] len 6 [Oct 10 14:06:18] DEBUG[15894] http.c: match request [ari/api-docs/asterisk.json] with handler [ari] len 3 [Oct 10 14:06:18] DEBUG[15894] http.c: Match made with [ari] [Oct 10 14:06:18] DEBUG[15894] res_ari.c: ast_ari_get_docs(asterisk.json) [Oct 10 14:06:18] DEBUG[15895] http.c: HTTP Request URI is /ari/api-docs/endpoints.json?api_key=admin:secret [Oct 10 14:06:18] DEBUG[15895] http.c: match request [ari/api-docs/endpoints.json] with handler [httpstatus] len 10 [Oct 10 14:06:18] DEBUG[15895] http.c: match request [ari/api-docs/endpoints.json] with handler [phoneprov] len 9 [Oct 10 14:06:18] DEBUG[15895] http.c: match request [ari/api-docs/endpoints.json] with handler [static] len 6 [Oct 10 14:06:18] DEBUG[15895] http.c: match request [ari/api-docs/endpoints.json] with handler [ari] len 3 [Oct 10 14:06:18] DEBUG[15895] http.c: Match made with [ari] [Oct 10 14:06:18] DEBUG[15895] res_ari.c: ast_ari_get_docs(endpoints.json) [Oct 10 14:06:18] DEBUG[15898] http.c: HTTP Request URI is /ari/api-docs/recordings.json?api_key=admin:secret [Oct 10 14:06:18] DEBUG[15898] http.c: match request [ari/api-docs/recordings.json] with handler [httpstatus] len 10 [Oct 10 14:06:18] DEBUG[15898] http.c: match request [ari/api-docs/recordings.json] with handler [phoneprov] len 9 [Oct 10 14:06:18] DEBUG[15898] http.c: match request [ari/api-docs/recordings.json] with handler [static] len 6 [Oct 10 14:06:18] DEBUG[15898] http.c: match request [ari/api-docs/recordings.json] with handler [ari] len 3 [Oct 10 14:06:18] DEBUG[15898] http.c: Match made with [ari] [Oct 10 14:06:18] DEBUG[15898] res_ari.c: ast_ari_get_docs(recordings.json) [Oct 10 14:06:18] DEBUG[15897] http.c: HTTP Request URI is /ari/api-docs/bridges.json?api_key=admin:secret [Oct 10 14:06:18] DEBUG[15897] http.c: match request [ari/api-docs/bridges.json] with handler [httpstatus] len 10 [Oct 10 14:06:18] DEBUG[15897] http.c: match request [ari/api-docs/bridges.json] with handler [phoneprov] len 9 [Oct 10 14:06:18] DEBUG[15897] http.c: match request [ari/api-docs/bridges.json] with handler [static] len 6 [Oct 10 14:06:18] DEBUG[15897] http.c: match request [ari/api-docs/bridges.json] with handler [ari] len 3 [Oct 10 14:06:18] DEBUG[15897] http.c: Match made with [ari] [Oct 10 14:06:18] DEBUG[15897] res_ari.c: ast_ari_get_docs(bridges.json) [Oct 10 14:06:18] DEBUG[15896] http.c: HTTP Request URI is /ari/api-docs/channels.json?api_key=admin:secret [Oct 10 14:06:18] DEBUG[15896] http.c: match request [ari/api-docs/channels.json] with handler [httpstatus] len 10 [Oct 10 14:06:18] DEBUG[15896] http.c: match request [ari/api-docs/channels.json] with handler [phoneprov] len 9 [Oct 10 14:06:18] DEBUG[15896] http.c: match request [ari/api-docs/channels.json] with handler [static] len 6 [Oct 10 14:06:18] DEBUG[15896] http.c: match request [ari/api-docs/channels.json] with handler [ari] len 3 [Oct 10 14:06:18] DEBUG[15896] http.c: Match made with [ari] [Oct 10 14:06:18] DEBUG[15896] res_ari.c: ast_ari_get_docs(channels.json) [Oct 10 14:06:18] DEBUG[15899] http.c: HTTP Request URI is /ari/api-docs/sounds.json?api_key=admin:secret [Oct 10 14:06:18] DEBUG[15899] http.c: match request [ari/api-docs/sounds.json] with handler [httpstatus] len 10 [Oct 10 14:06:18] DEBUG[15899] http.c: match request [ari/api-docs/sounds.json] with handler [phoneprov] len 9 [Oct 10 14:06:18] DEBUG[15899] http.c: match request [ari/api-docs/sounds.json] with handler [static] len 6 [Oct 10 14:06:18] DEBUG[15899] http.c: match request [ari/api-docs/sounds.json] with handler [ari] len 3 [Oct 10 14:06:18] DEBUG[15899] http.c: Match made with [ari] [Oct 10 14:06:18] DEBUG[15899] res_ari.c: ast_ari_get_docs(sounds.json) [Oct 10 14:06:18] DEBUG[15901] http.c: HTTP Request URI is /ari/api-docs/events.json?api_key=admin:secret [Oct 10 14:06:18] DEBUG[15901] http.c: match request [ari/api-docs/events.json] with handler [httpstatus] len 10 [Oct 10 14:06:18] DEBUG[15901] http.c: match request [ari/api-docs/events.json] with handler [phoneprov] len 9 [Oct 10 14:06:18] DEBUG[15901] http.c: match request [ari/api-docs/events.json] with handler [static] len 6 [Oct 10 14:06:18] DEBUG[15901] http.c: match request [ari/api-docs/events.json] with handler [ari] len 3 [Oct 10 14:06:18] DEBUG[15901] http.c: Match made with [ari] [Oct 10 14:06:18] DEBUG[15901] res_ari.c: ast_ari_get_docs(events.json) [Oct 10 14:06:18] DEBUG[15900] http.c: HTTP Request URI is /ari/api-docs/playback.json?api_key=admin:secret [Oct 10 14:06:18] DEBUG[15900] http.c: match request [ari/api-docs/playback.json] with handler [httpstatus] len 10 [Oct 10 14:06:18] DEBUG[15900] http.c: match request [ari/api-docs/playback.json] with handler [phoneprov] len 9 [Oct 10 14:06:18] DEBUG[15900] http.c: match request [ari/api-docs/playback.json] with handler [static] len 6 [Oct 10 14:06:18] DEBUG[15900] http.c: match request [ari/api-docs/playback.json] with handler [ari] len 3 [Oct 10 14:06:18] DEBUG[15900] http.c: Match made with [ari] [Oct 10 14:06:18] DEBUG[15900] res_ari.c: ast_ari_get_docs(playback.json) [Oct 10 14:06:18] DEBUG[15902] http.c: HTTP Request URI is /ari/api-docs/applications.json?api_key=admin:secret [Oct 10 14:06:18] DEBUG[15902] http.c: match request [ari/api-docs/applications.json] with handler [httpstatus] len 10 [Oct 10 14:06:18] DEBUG[15902] http.c: match request [ari/api-docs/applications.json] with handler [phoneprov] len 9 [Oct 10 14:06:18] DEBUG[15902] http.c: match request [ari/api-docs/applications.json] with handler [static] len 6 [Oct 10 14:06:18] DEBUG[15902] http.c: match request [ari/api-docs/applications.json] with handler [ari] len 3 [Oct 10 14:06:18] DEBUG[15902] http.c: Match made with [ari] [Oct 10 14:06:18] DEBUG[15902] res_ari.c: ast_ari_get_docs(applications.json) [Oct 10 14:06:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:06:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:06:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:24] DEBUG[15903] http.c: HTTP Request URI is /ari/channels?api_key=admin%3Asecret [Oct 10 14:06:24] DEBUG[15903] http.c: match request [ari/channels] with handler [httpstatus] len 10 [Oct 10 14:06:24] DEBUG[15903] http.c: match request [ari/channels] with handler [phoneprov] len 9 [Oct 10 14:06:24] DEBUG[15903] http.c: match request [ari/channels] with handler [static] len 6 [Oct 10 14:06:24] DEBUG[15903] http.c: match request [ari/channels] with handler [ari] len 3 [Oct 10 14:06:24] DEBUG[15903] http.c: Match made with [ari] [Oct 10 14:06:24] DEBUG[15903] res_ari.c: Finding handler for channels [Oct 10 14:06:24] DEBUG[15903] res_ari.c: Checking endpoints [Oct 10 14:06:24] DEBUG[15903] res_ari.c: Checking channels [Oct 10 14:06:24] DEBUG[15903] res_ari.c: Got it! [Oct 10 14:06:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:48] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:06:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:22] DEBUG[15908] http.c: HTTP Request URI is /ari/api-docs/resources.json?api_key=admin:secret [Oct 10 14:07:22] DEBUG[15908] http.c: match request [ari/api-docs/resources.json] with handler [httpstatus] len 10 [Oct 10 14:07:22] DEBUG[15908] http.c: match request [ari/api-docs/resources.json] with handler [phoneprov] len 9 [Oct 10 14:07:22] DEBUG[15908] http.c: match request [ari/api-docs/resources.json] with handler [static] len 6 [Oct 10 14:07:22] DEBUG[15908] http.c: match request [ari/api-docs/resources.json] with handler [ari] len 3 [Oct 10 14:07:22] DEBUG[15908] http.c: Match made with [ari] [Oct 10 14:07:22] DEBUG[15908] res_ari.c: ast_ari_get_docs(resources.json) [Oct 10 14:07:22] DEBUG[15909] http.c: HTTP Request URI is /ari/api-docs/asterisk.json?api_key=admin:secret [Oct 10 14:07:22] DEBUG[15909] http.c: match request [ari/api-docs/asterisk.json] with handler [httpstatus] len 10 [Oct 10 14:07:22] DEBUG[15909] http.c: match request [ari/api-docs/asterisk.json] with handler [phoneprov] len 9 [Oct 10 14:07:22] DEBUG[15909] http.c: match request [ari/api-docs/asterisk.json] with handler [static] len 6 [Oct 10 14:07:22] DEBUG[15909] http.c: match request [ari/api-docs/asterisk.json] with handler [ari] len 3 [Oct 10 14:07:22] DEBUG[15909] http.c: Match made with [ari] [Oct 10 14:07:22] DEBUG[15909] res_ari.c: ast_ari_get_docs(asterisk.json) [Oct 10 14:07:22] DEBUG[15912] http.c: HTTP Request URI is /ari/api-docs/bridges.json?api_key=admin:secret [Oct 10 14:07:22] DEBUG[15912] http.c: match request [ari/api-docs/bridges.json] with handler [httpstatus] len 10 [Oct 10 14:07:22] DEBUG[15913] http.c: HTTP Request URI is /ari/api-docs/recordings.json?api_key=admin:secret [Oct 10 14:07:22] DEBUG[15912] http.c: match request [ari/api-docs/bridges.json] with handler [phoneprov] len 9 [Oct 10 14:07:22] DEBUG[15912] http.c: match request [ari/api-docs/bridges.json] with handler [static] len 6 [Oct 10 14:07:22] DEBUG[15913] http.c: match request [ari/api-docs/recordings.json] with handler [httpstatus] len 10 [Oct 10 14:07:22] DEBUG[15913] http.c: match request [ari/api-docs/recordings.json] with handler [phoneprov] len 9 [Oct 10 14:07:22] DEBUG[15913] http.c: match request [ari/api-docs/recordings.json] with handler [static] len 6 [Oct 10 14:07:22] DEBUG[15913] http.c: match request [ari/api-docs/recordings.json] with handler [ari] len 3 [Oct 10 14:07:22] DEBUG[15913] http.c: Match made with [ari] [Oct 10 14:07:22] DEBUG[15910] http.c: HTTP Request URI is /ari/api-docs/endpoints.json?api_key=admin:secret [Oct 10 14:07:22] DEBUG[15910] http.c: match request [ari/api-docs/endpoints.json] with handler [httpstatus] len 10 [Oct 10 14:07:22] DEBUG[15910] http.c: match request [ari/api-docs/endpoints.json] with handler [phoneprov] len 9 [Oct 10 14:07:22] DEBUG[15910] http.c: match request [ari/api-docs/endpoints.json] with handler [static] len 6 [Oct 10 14:07:22] DEBUG[15910] http.c: match request [ari/api-docs/endpoints.json] with handler [ari] len 3 [Oct 10 14:07:22] DEBUG[15910] http.c: Match made with [ari] [Oct 10 14:07:22] DEBUG[15910] res_ari.c: ast_ari_get_docs(endpoints.json) [Oct 10 14:07:22] DEBUG[15912] http.c: match request [ari/api-docs/bridges.json] with handler [ari] len 3 [Oct 10 14:07:22] DEBUG[15912] http.c: Match made with [ari] [Oct 10 14:07:22] DEBUG[15912] res_ari.c: ast_ari_get_docs(bridges.json) [Oct 10 14:07:22] DEBUG[15913] res_ari.c: ast_ari_get_docs(recordings.json) [Oct 10 14:07:22] DEBUG[15911] http.c: HTTP Request URI is /ari/api-docs/channels.json?api_key=admin:secret [Oct 10 14:07:22] DEBUG[15911] http.c: match request [ari/api-docs/channels.json] with handler [httpstatus] len 10 [Oct 10 14:07:22] DEBUG[15911] http.c: match request [ari/api-docs/channels.json] with handler [phoneprov] len 9 [Oct 10 14:07:22] DEBUG[15911] http.c: match request [ari/api-docs/channels.json] with handler [static] len 6 [Oct 10 14:07:22] DEBUG[15911] http.c: match request [ari/api-docs/channels.json] with handler [ari] len 3 [Oct 10 14:07:22] DEBUG[15911] http.c: Match made with [ari] [Oct 10 14:07:22] DEBUG[15911] res_ari.c: ast_ari_get_docs(channels.json) [Oct 10 14:07:22] DEBUG[15914] http.c: HTTP Request URI is /ari/api-docs/sounds.json?api_key=admin:secret [Oct 10 14:07:22] DEBUG[15914] http.c: match request [ari/api-docs/sounds.json] with handler [httpstatus] len 10 [Oct 10 14:07:22] DEBUG[15914] http.c: match request [ari/api-docs/sounds.json] with handler [phoneprov] len 9 [Oct 10 14:07:22] DEBUG[15914] http.c: match request [ari/api-docs/sounds.json] with handler [static] len 6 [Oct 10 14:07:22] DEBUG[15914] http.c: match request [ari/api-docs/sounds.json] with handler [ari] len 3 [Oct 10 14:07:22] DEBUG[15914] http.c: Match made with [ari] [Oct 10 14:07:22] DEBUG[15914] res_ari.c: ast_ari_get_docs(sounds.json) [Oct 10 14:07:22] DEBUG[15916] http.c: HTTP Request URI is /ari/api-docs/playback.json?api_key=admin:secret [Oct 10 14:07:22] DEBUG[15916] http.c: match request [ari/api-docs/playback.json] with handler [httpstatus] len 10 [Oct 10 14:07:22] DEBUG[15916] http.c: match request [ari/api-docs/playback.json] with handler [phoneprov] len 9 [Oct 10 14:07:22] DEBUG[15916] http.c: match request [ari/api-docs/playback.json] with handler [static] len 6 [Oct 10 14:07:22] DEBUG[15916] http.c: match request [ari/api-docs/playback.json] with handler [ari] len 3 [Oct 10 14:07:22] DEBUG[15916] http.c: Match made with [ari] [Oct 10 14:07:22] DEBUG[15915] http.c: HTTP Request URI is /ari/api-docs/events.json?api_key=admin:secret [Oct 10 14:07:22] DEBUG[15915] http.c: match request [ari/api-docs/events.json] with handler [httpstatus] len 10 [Oct 10 14:07:22] DEBUG[15915] http.c: match request [ari/api-docs/events.json] with handler [phoneprov] len 9 [Oct 10 14:07:22] DEBUG[15915] http.c: match request [ari/api-docs/events.json] with handler [static] len 6 [Oct 10 14:07:22] DEBUG[15915] http.c: match request [ari/api-docs/events.json] with handler [ari] len 3 [Oct 10 14:07:22] DEBUG[15915] http.c: Match made with [ari] [Oct 10 14:07:22] DEBUG[15915] res_ari.c: ast_ari_get_docs(events.json) [Oct 10 14:07:22] DEBUG[15916] res_ari.c: ast_ari_get_docs(playback.json) [Oct 10 14:07:22] DEBUG[15917] http.c: HTTP Request URI is /ari/api-docs/applications.json?api_key=admin:secret [Oct 10 14:07:22] DEBUG[15917] http.c: match request [ari/api-docs/applications.json] with handler [httpstatus] len 10 [Oct 10 14:07:22] DEBUG[15917] http.c: match request [ari/api-docs/applications.json] with handler [phoneprov] len 9 [Oct 10 14:07:22] DEBUG[15917] http.c: match request [ari/api-docs/applications.json] with handler [static] len 6 [Oct 10 14:07:22] DEBUG[15917] http.c: match request [ari/api-docs/applications.json] with handler [ari] len 3 [Oct 10 14:07:22] DEBUG[15917] http.c: Match made with [ari] [Oct 10 14:07:22] DEBUG[15917] res_ari.c: ast_ari_get_docs(applications.json) [Oct 10 14:07:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:27] DEBUG[15918] http.c: HTTP Request URI is /ari/channels?api_key=admin%3Asecret [Oct 10 14:07:27] DEBUG[15918] http.c: match request [ari/channels] with handler [httpstatus] len 10 [Oct 10 14:07:27] DEBUG[15918] http.c: match request [ari/channels] with handler [phoneprov] len 9 [Oct 10 14:07:27] DEBUG[15918] http.c: match request [ari/channels] with handler [static] len 6 [Oct 10 14:07:27] DEBUG[15918] http.c: match request [ari/channels] with handler [ari] len 3 [Oct 10 14:07:27] DEBUG[15918] http.c: Match made with [ari] [Oct 10 14:07:27] DEBUG[15918] res_ari.c: Finding handler for channels [Oct 10 14:07:27] DEBUG[15918] res_ari.c: Checking endpoints [Oct 10 14:07:27] DEBUG[15918] res_ari.c: Checking channels [Oct 10 14:07:27] DEBUG[15918] res_ari.c: Got it! [Oct 10 14:07:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:48] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:07:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:07:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:07:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:07:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:07:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:07:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:07:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:07:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for C.tDxR6v9c6WOlPO2krfnaJ6rts3NBBZ - SUBSCRIBE (No RTP) [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:07:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:07:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog C.tDxR6v9c6WOlPO2krfnaJ6rts3NBBZ [Oct 10 14:07:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:07:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:08:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:08:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:08:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:08:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:08:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:08:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:08:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:08:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for WVA90nPmJu5j0vjyVKLvqKXLKyPbkLkU - SUBSCRIBE (No RTP) [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:08:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:08:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog WVA90nPmJu5j0vjyVKLvqKXLKyPbkLkU [Oct 10 14:08:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:08:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:08:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:44] DEBUG[15924] http.c: HTTP Request URI is /ari/channels?api_key=admin:secret [Oct 10 14:08:44] DEBUG[15924] http.c: match request [ari/channels] with handler [httpstatus] len 10 [Oct 10 14:08:44] DEBUG[15924] http.c: match request [ari/channels] with handler [phoneprov] len 9 [Oct 10 14:08:44] DEBUG[15924] http.c: match request [ari/channels] with handler [static] len 6 [Oct 10 14:08:44] DEBUG[15924] http.c: match request [ari/channels] with handler [ari] len 3 [Oct 10 14:08:44] DEBUG[15924] http.c: Match made with [ari] [Oct 10 14:08:44] DEBUG[15924] res_ari.c: Finding handler for channels [Oct 10 14:08:44] DEBUG[15924] res_ari.c: Checking endpoints [Oct 10 14:08:44] DEBUG[15924] res_ari.c: Checking channels [Oct 10 14:08:44] DEBUG[15924] res_ari.c: Got it! [Oct 10 14:08:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:48] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:08:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:08:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:08:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:08:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for nzlMLw2zi-8Ij-bgpPGzY3d6EvGKPc0K - SUBSCRIBE (No RTP) [Oct 10 14:08:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:08:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:08:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:08:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:08:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog nzlMLw2zi-8Ij-bgpPGzY3d6EvGKPc0K [Oct 10 14:08:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:08:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:08] DEBUG[15928] http.c: HTTP Request URI is /ari/bridges?api_key=admin%3Asecret [Oct 10 14:09:08] DEBUG[15928] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:09:08] DEBUG[15928] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:09:08] DEBUG[15928] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:09:08] DEBUG[15928] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:09:08] DEBUG[15928] http.c: Match made with [ari] [Oct 10 14:09:08] DEBUG[15928] res_ari.c: Finding handler for bridges [Oct 10 14:09:08] DEBUG[15928] res_ari.c: Checking endpoints [Oct 10 14:09:08] DEBUG[15928] res_ari.c: Checking channels [Oct 10 14:09:08] DEBUG[15928] res_ari.c: Checking events [Oct 10 14:09:08] DEBUG[15928] res_ari.c: Checking recordings [Oct 10 14:09:08] DEBUG[15928] res_ari.c: Checking playback [Oct 10 14:09:08] DEBUG[15928] res_ari.c: Checking applications [Oct 10 14:09:08] DEBUG[15928] res_ari.c: Checking bridges [Oct 10 14:09:08] DEBUG[15928] res_ari.c: Got it! [Oct 10 14:09:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:28] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:09:28] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:09:28] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:09:28] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:09:28] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:09:28] DEBUG[15619] chan_sip.c: Acked pending invite 103 [Oct 10 14:09:28] DEBUG[15619] chan_sip.c: Stopping retransmission on 'PLockzot5wXYYBQ89eWd18qPrTFlxMPl' of Request 103: Match Found [Oct 10 14:09:28] DEBUG[15930] http.c: HTTP Request URI is /ari/bridges?api_key=admin:secret [Oct 10 14:09:28] DEBUG[15930] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:09:28] DEBUG[15930] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:09:28] DEBUG[15930] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:09:28] DEBUG[15930] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:09:28] DEBUG[15930] http.c: Match made with [ari] [Oct 10 14:09:28] DEBUG[15930] res_ari.c: Finding handler for bridges [Oct 10 14:09:28] DEBUG[15930] res_ari.c: Checking endpoints [Oct 10 14:09:28] DEBUG[15930] res_ari.c: Checking channels [Oct 10 14:09:28] DEBUG[15930] res_ari.c: Checking events [Oct 10 14:09:28] DEBUG[15930] res_ari.c: Checking recordings [Oct 10 14:09:28] DEBUG[15930] res_ari.c: Checking playback [Oct 10 14:09:28] DEBUG[15930] res_ari.c: Checking applications [Oct 10 14:09:28] DEBUG[15930] res_ari.c: Checking bridges [Oct 10 14:09:28] DEBUG[15930] res_ari.c: Got it! [Oct 10 14:09:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:44] DEBUG[15932] http.c: HTTP Request URI is /ari/bridges?type=holding&api_key=admin%3Asecret [Oct 10 14:09:44] DEBUG[15932] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:09:44] DEBUG[15932] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:09:44] DEBUG[15932] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:09:44] DEBUG[15932] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:09:44] DEBUG[15932] http.c: Match made with [ari] [Oct 10 14:09:44] DEBUG[15932] res_ari.c: Finding handler for bridges [Oct 10 14:09:44] DEBUG[15932] res_ari.c: Checking endpoints [Oct 10 14:09:44] DEBUG[15932] res_ari.c: Checking channels [Oct 10 14:09:44] DEBUG[15932] res_ari.c: Checking events [Oct 10 14:09:44] DEBUG[15932] res_ari.c: Checking recordings [Oct 10 14:09:44] DEBUG[15932] res_ari.c: Checking playback [Oct 10 14:09:44] DEBUG[15932] res_ari.c: Checking applications [Oct 10 14:09:44] DEBUG[15932] res_ari.c: Checking bridges [Oct 10 14:09:44] DEBUG[15932] res_ari.c: Got it! [Oct 10 14:09:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:48] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:09:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:09:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 31k9abPxFDNPOAYCWzGYyKuOkzs-pZzt - SUBSCRIBE (No RTP) [Oct 10 14:09:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:09:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:09:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:09:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:09:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog 31k9abPxFDNPOAYCWzGYyKuOkzs-pZzt [Oct 10 14:09:50] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:09:50] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:09:50] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:09:50] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:09:50] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:09:50] DEBUG[15619] chan_sip.c: Acked pending invite 103 [Oct 10 14:09:50] DEBUG[15619] chan_sip.c: Stopping retransmission on 'EVj1DqKxYCA3vR-L.ceVXGwe5iq0.Ixi' of Request 103: Match Found [Oct 10 14:09:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:09:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:07] DEBUG[15936] http.c: HTTP Request URI is /ari/bridges?type=holding&api_key=admin:secret [Oct 10 14:10:07] DEBUG[15936] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:10:07] DEBUG[15936] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:10:07] DEBUG[15936] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:10:07] DEBUG[15936] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:10:07] DEBUG[15936] http.c: Match made with [ari] [Oct 10 14:10:07] DEBUG[15936] res_ari.c: Finding handler for bridges [Oct 10 14:10:07] DEBUG[15936] res_ari.c: Checking endpoints [Oct 10 14:10:07] DEBUG[15936] res_ari.c: Checking channels [Oct 10 14:10:07] DEBUG[15936] res_ari.c: Checking events [Oct 10 14:10:07] DEBUG[15936] res_ari.c: Checking recordings [Oct 10 14:10:07] DEBUG[15936] res_ari.c: Checking playback [Oct 10 14:10:07] DEBUG[15936] res_ari.c: Checking applications [Oct 10 14:10:07] DEBUG[15936] res_ari.c: Checking bridges [Oct 10 14:10:07] DEBUG[15936] res_ari.c: Got it! [Oct 10 14:10:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:22] DEBUG[15938] http.c: HTTP Request URI is /ari/bridges?type=holding&api_key=admin:secret [Oct 10 14:10:22] DEBUG[15938] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:10:22] DEBUG[15938] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:10:22] DEBUG[15938] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:10:22] DEBUG[15938] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:10:22] DEBUG[15938] http.c: Match made with [ari] [Oct 10 14:10:22] DEBUG[15938] res_ari.c: Finding handler for bridges [Oct 10 14:10:22] DEBUG[15938] res_ari.c: Checking endpoints [Oct 10 14:10:22] DEBUG[15938] res_ari.c: Checking channels [Oct 10 14:10:22] DEBUG[15938] res_ari.c: Checking events [Oct 10 14:10:22] DEBUG[15938] res_ari.c: Checking recordings [Oct 10 14:10:22] DEBUG[15938] res_ari.c: Checking playback [Oct 10 14:10:22] DEBUG[15938] res_ari.c: Checking applications [Oct 10 14:10:22] DEBUG[15938] res_ari.c: Checking bridges [Oct 10 14:10:22] DEBUG[15938] res_ari.c: Got it! [Oct 10 14:10:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:10:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:10:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:10:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:10:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:10:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:10:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:10:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:10:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for eXQeWhhlaCeZJmcS3hNYnRwTCTtyFAq1 - SUBSCRIBE (No RTP) [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:10:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:10:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog eXQeWhhlaCeZJmcS3hNYnRwTCTtyFAq1 [Oct 10 14:10:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:29] DEBUG[15940] http.c: HTTP Request URI is /ari/bridges?type=holding&api_key=admin:secret [Oct 10 14:10:29] DEBUG[15940] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:10:29] DEBUG[15940] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:10:29] DEBUG[15940] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:10:29] DEBUG[15940] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:10:29] DEBUG[15940] http.c: Match made with [ari] [Oct 10 14:10:29] DEBUG[15940] res_ari.c: Finding handler for bridges [Oct 10 14:10:29] DEBUG[15940] res_ari.c: Checking endpoints [Oct 10 14:10:29] DEBUG[15940] res_ari.c: Checking channels [Oct 10 14:10:29] DEBUG[15940] res_ari.c: Checking events [Oct 10 14:10:29] DEBUG[15940] res_ari.c: Checking recordings [Oct 10 14:10:29] DEBUG[15940] res_ari.c: Checking playback [Oct 10 14:10:29] DEBUG[15940] res_ari.c: Checking applications [Oct 10 14:10:29] DEBUG[15940] res_ari.c: Checking bridges [Oct 10 14:10:29] DEBUG[15940] res_ari.c: Got it! [Oct 10 14:10:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:10:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:10:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:10:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:10:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:10:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:10:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:10:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:10:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for h.0DL4srffJ3jnd4CKDaN.lzuBOvKwWb - SUBSCRIBE (No RTP) [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:10:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:10:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog h.0DL4srffJ3jnd4CKDaN.lzuBOvKwWb [Oct 10 14:10:48] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:10:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:10:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:10:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:11:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:11:20] DEBUG[15944] http.c: HTTP Request URI is /ari/bridges?api_key=admin%3Asecret [Oct 10 14:11:20] DEBUG[15944] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:11:20] DEBUG[15944] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:11:20] DEBUG[15944] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:11:20] DEBUG[15944] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:11:20] DEBUG[15944] http.c: Match made with [ari] [Oct 10 14:11:20] DEBUG[15944] res_ari.c: Finding handler for bridges [Oct 10 14:11:20] DEBUG[15944] res_ari.c: Checking endpoints [Oct 10 14:11:20] DEBUG[15944] res_ari.c: Checking channels [Oct 10 14:11:20] DEBUG[15944] res_ari.c: Checking events [Oct 10 14:11:20] DEBUG[15944] res_ari.c: Checking recordings [Oct 10 14:11:20] DEBUG[15944] res_ari.c: Checking playback [Oct 10 14:11:20] DEBUG[15944] res_ari.c: Checking applications [Oct 10 14:11:20] DEBUG[15944] res_ari.c: Checking bridges [Oct 10 14:11:20] DEBUG[15944] res_ari.c: Got it! [Oct 10 14:11:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:48] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:11:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:01] DEBUG[15948] http.c: HTTP Request URI is /ari/bridges?type=holding&operation=create&api_key=admin:secret [Oct 10 14:12:01] DEBUG[15948] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:12:01] DEBUG[15948] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:12:01] DEBUG[15948] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:12:01] DEBUG[15948] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:12:01] DEBUG[15948] http.c: Match made with [ari] [Oct 10 14:12:01] DEBUG[15948] res_ari.c: Finding handler for bridges [Oct 10 14:12:01] DEBUG[15948] res_ari.c: Checking endpoints [Oct 10 14:12:01] DEBUG[15948] res_ari.c: Checking channels [Oct 10 14:12:01] DEBUG[15948] res_ari.c: Checking events [Oct 10 14:12:01] DEBUG[15948] res_ari.c: Checking recordings [Oct 10 14:12:01] DEBUG[15948] res_ari.c: Checking playback [Oct 10 14:12:01] DEBUG[15948] res_ari.c: Checking applications [Oct 10 14:12:01] DEBUG[15948] res_ari.c: Checking bridges [Oct 10 14:12:01] DEBUG[15948] res_ari.c: Got it! [Oct 10 14:12:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:45] DEBUG[15952] http.c: HTTP Request URI is /ari/bridges?type=holding&operation=create&api_key=admin:secret [Oct 10 14:12:45] DEBUG[15952] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:12:45] DEBUG[15952] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:12:45] DEBUG[15952] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:12:45] DEBUG[15952] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:12:45] DEBUG[15952] http.c: Match made with [ari] [Oct 10 14:12:45] DEBUG[15952] res_ari.c: Finding handler for bridges [Oct 10 14:12:45] DEBUG[15952] res_ari.c: Checking endpoints [Oct 10 14:12:45] DEBUG[15952] res_ari.c: Checking channels [Oct 10 14:12:45] DEBUG[15952] res_ari.c: Checking events [Oct 10 14:12:45] DEBUG[15952] res_ari.c: Checking recordings [Oct 10 14:12:45] DEBUG[15952] res_ari.c: Checking playback [Oct 10 14:12:45] DEBUG[15952] res_ari.c: Checking applications [Oct 10 14:12:45] DEBUG[15952] res_ari.c: Checking bridges [Oct 10 14:12:45] DEBUG[15952] res_ari.c: Got it! [Oct 10 14:12:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:53] DEBUG[15954] http.c: HTTP Request URI is /ari/bridges?type=holding&api_key=admin:secret [Oct 10 14:12:53] DEBUG[15954] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:12:53] DEBUG[15954] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:12:53] DEBUG[15954] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:12:53] DEBUG[15954] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:12:53] DEBUG[15954] http.c: Match made with [ari] [Oct 10 14:12:53] DEBUG[15954] res_ari.c: Finding handler for bridges [Oct 10 14:12:53] DEBUG[15954] res_ari.c: Checking endpoints [Oct 10 14:12:53] DEBUG[15954] res_ari.c: Checking channels [Oct 10 14:12:53] DEBUG[15954] res_ari.c: Checking events [Oct 10 14:12:53] DEBUG[15954] res_ari.c: Checking recordings [Oct 10 14:12:53] DEBUG[15954] res_ari.c: Checking playback [Oct 10 14:12:53] DEBUG[15954] res_ari.c: Checking applications [Oct 10 14:12:53] DEBUG[15954] res_ari.c: Checking bridges [Oct 10 14:12:53] DEBUG[15954] res_ari.c: Got it! [Oct 10 14:12:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:12:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:12:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:12:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:12:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:12:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:12:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:12:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:12:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for lWYXOlgTDIJSZDpOLB5bHo6hORUgAH6T - SUBSCRIBE (No RTP) [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:12:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:12:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog lWYXOlgTDIJSZDpOLB5bHo6hORUgAH6T [Oct 10 14:12:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:12:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:13:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:13:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:13:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:13:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:13:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:13:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:13:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:13:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for r3w2z6rseUjPnbUYpblX3sWNgICu8P4v - SUBSCRIBE (No RTP) [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:13:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:13:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog r3w2z6rseUjPnbUYpblX3sWNgICu8P4v [Oct 10 14:13:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:13:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:13:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:13:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:13:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:13:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:13:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for AFpszJGGEtsjkGvKhapS-IR4aiqmfDfk - SUBSCRIBE (No RTP) [Oct 10 14:13:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:13:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:13:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:13:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:13:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog AFpszJGGEtsjkGvKhapS-IR4aiqmfDfk [Oct 10 14:13:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:13:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:14:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:14:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 7To-I8Ao01055n6akfenyHr1Nj245VPF - SUBSCRIBE (No RTP) [Oct 10 14:14:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:14:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:14:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:14:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:14:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog 7To-I8Ao01055n6akfenyHr1Nj245VPF [Oct 10 14:14:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:14:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:15:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:15:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:15:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:15:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:15:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:15:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:15:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:15:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 43QAOcviNB7hwkLkGf6SUK4TKuPot6-e - SUBSCRIBE (No RTP) [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:15:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:15:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog 43QAOcviNB7hwkLkGf6SUK4TKuPot6-e [Oct 10 14:15:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:15:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:15:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:15:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:15:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:15:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:15:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:15:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:15:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for GQdTvsGSs-7eju9-2AzNaNHqpMj9JlEK - SUBSCRIBE (No RTP) [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:15:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:15:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog GQdTvsGSs-7eju9-2AzNaNHqpMj9JlEK [Oct 10 14:15:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:15:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:15:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:15:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:16:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:16:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:16:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:30] DEBUG[15992] http.c: HTTP Request URI is /ari/bridges?&api_key=admin:secret [Oct 10 14:17:30] DEBUG[15992] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:17:30] DEBUG[15992] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:17:30] DEBUG[15992] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:17:30] DEBUG[15992] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:17:30] DEBUG[15992] http.c: Match made with [ari] [Oct 10 14:17:30] DEBUG[15992] res_ari.c: Finding handler for bridges [Oct 10 14:17:30] DEBUG[15992] res_ari.c: Checking endpoints [Oct 10 14:17:30] DEBUG[15992] res_ari.c: Checking channels [Oct 10 14:17:30] DEBUG[15992] res_ari.c: Checking events [Oct 10 14:17:30] DEBUG[15992] res_ari.c: Checking recordings [Oct 10 14:17:30] DEBUG[15992] res_ari.c: Checking playback [Oct 10 14:17:30] DEBUG[15992] res_ari.c: Checking applications [Oct 10 14:17:30] DEBUG[15992] res_ari.c: Checking bridges [Oct 10 14:17:30] DEBUG[15992] res_ari.c: Got it! [Oct 10 14:17:30] DEBUG[15992] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Must have two channels. [Oct 10 14:17:30] DEBUG[15992] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:17:30] DEBUG[15992] bridge_native_rtp.c: Bridge 'f7c9433b-711e-4fdf-990e-7ea806eed848' can not use native RTP bridge as two channels are required [Oct 10 14:17:30] DEBUG[15992] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 10 14:17:30] DEBUG[15992] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:17:30] DEBUG[15992] bridge.c: Chose bridge technology simple_bridge [Oct 10 14:17:30] DEBUG[15992] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling simple_bridge technology constructor [Oct 10 14:17:30] DEBUG[15992] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling simple_bridge technology start [Oct 10 14:17:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:17:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:17:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:17:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:17:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:17:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:17:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:17:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:17:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QNMkv9Pur2EM2vMKY55b3IQYGn6knIqx - SUBSCRIBE (No RTP) [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:17:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:17:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog QNMkv9Pur2EM2vMKY55b3IQYGn6knIqx [Oct 10 14:17:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:17:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:18:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:18:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:18:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:18:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:18:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:18:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:18:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:18:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for fEtbcKy-kmTSghvMSohV-mDEUKVcSpcL - SUBSCRIBE (No RTP) [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:18:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:18:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog fEtbcKy-kmTSghvMSohV-mDEUKVcSpcL [Oct 10 14:18:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:18:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:18:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:18:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:18:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:18:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:18:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for J8LpnwfSninxVDsBdxkoe2gR9nkHkGhN - SUBSCRIBE (No RTP) [Oct 10 14:18:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:18:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:18:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:18:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:18:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog J8LpnwfSninxVDsBdxkoe2gR9nkHkGhN [Oct 10 14:18:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:18:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:23] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:19:23] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:19:23] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:19:23] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:19:23] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:19:23] DEBUG[15619] chan_sip.c: Acked pending invite 104 [Oct 10 14:19:23] DEBUG[15619] chan_sip.c: Stopping retransmission on 'PLockzot5wXYYBQ89eWd18qPrTFlxMPl' of Request 104: Match Found [Oct 10 14:19:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:33] DEBUG[15619] acl.c: For destination '10.24.17.17', our source address is '10.24.18.161'. [Oct 10 14:19:33] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:19:33] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 386d43afbe1b-td1bumlsie3r - REGISTER (No RTP) [Oct 10 14:19:33] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:19:33] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:19:33] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:19:33] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.17.17:5060 [Oct 10 14:19:33] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_C [Oct 10 14:19:33] DEBUG[15569] chan_sip.c: Checking device state for peer phone_C [Oct 10 14:19:33] DEBUG[15569] devicestate.c: Changing state for SIP/phone_C - state 1 (Not in use) [Oct 10 14:19:33] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_C' [Oct 10 14:19:33] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_C' [Oct 10 14:19:33] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_C' [Oct 10 14:19:33] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_C' has changed to 'Not in use' [Oct 10 14:19:33] DEBUG[15681] app_queue.c: Device 'SIP/phone_C' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Oct 10 14:19:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:44] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:19:44] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:19:44] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:19:44] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:19:44] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:19:45] DEBUG[15619] chan_sip.c: Acked pending invite 104 [Oct 10 14:19:45] DEBUG[15619] chan_sip.c: Stopping retransmission on 'EVj1DqKxYCA3vR-L.ceVXGwe5iq0.Ixi' of Request 104: Match Found [Oct 10 14:19:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:19:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:19:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for DhbJp7oXw4vbfvWh.QrGiKk4VY88ARaZ - SUBSCRIBE (No RTP) [Oct 10 14:19:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:19:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:19:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:19:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:19:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog DhbJp7oXw4vbfvWh.QrGiKk4VY88ARaZ [Oct 10 14:19:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:19:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:05] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog '386d43afbe1b-td1bumlsie3r' [Oct 10 14:20:05] DEBUG[15619] chan_sip.c: Destroying SIP dialog 386d43afbe1b-td1bumlsie3r [Oct 10 14:20:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:20:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:20:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:20:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:20:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:20:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:20:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:20:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:20:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for bT5vZ.LTDj9adOmyAx.WM6jMWr0LUDnx - SUBSCRIBE (No RTP) [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:20:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:20:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog bT5vZ.LTDj9adOmyAx.WM6jMWr0LUDnx [Oct 10 14:20:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:43] DEBUG[15619] chan_sip.c: Session timer expired: 44 - iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX [Oct 10 14:20:43] DEBUG[15619] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Oct 10 14:20:43] DEBUG[15619] chan_sip.c: ** Our prefcodec: (nothing) [Oct 10 14:20:43] DEBUG[15619] chan_sip.c: -- Done with adding codecs to SDP [Oct 10 14:20:43] DEBUG[15619] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Oct 10 14:20:43] DEBUG[15619] chan_sip.c: Initializing already initialized SIP dialog iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX (presumably reinvite) [Oct 10 14:20:43] DEBUG[15619] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:20:43] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 48 bytes [Oct 10 14:20:43] DEBUG[15619][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Acked pending invite 102 [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Stopping retransmission on 'iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX' of Request 102: Match Found [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: SIP response 200 to RE-invite on outgoing call iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP o=- 76803233 76803234 IN IP4 10.24.19.97... OK. [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.19.97... OK. [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Oct 10 14:20:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0xb2d91ca8 [Oct 10 14:20:43] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0xb2d91ca8 [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4031 IN IP4 10.24.19.97... UNSUPPORTED OR FAILED. [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Oct 10 14:20:43] DEBUG[15619][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb2a0a374' [Oct 10 14:20:43] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 0 from 0xb2d91ca8 to 0xb2a0a49c [Oct 10 14:20:43] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 96 from 0xb2d91ca8 to 0xb2a0a49c [Oct 10 14:20:43] DEBUG[15619][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb2a0a374' [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: We have an owner, now see if we need to change this call [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Updating call counter for incoming call [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Session-Expires: 1800 [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Refresher: UAC [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Session timer stopped: 44 - iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Session timer started: 87 - iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX 900000ms [Oct 10 14:20:43] DEBUG[15619][C-00000001] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:20:43] DEBUG[15619][C-00000001] logger.c: CALL_ID [C-00000001] being removed from thread. [Oct 10 14:20:43] DEBUG[15888][C-00000001] res_rtp_asterisk.c: 0xb2cc9aa8 -- Probation learning mode pass with source address 10.24.19.97:4030 [Oct 10 14:20:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:20:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:20:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:20:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:20:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:20:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:20:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:20:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:20:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for f21vrt.8TItvrm3Vz6sv-G5boJOowl8n - SUBSCRIBE (No RTP) [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:20:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:20:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog f21vrt.8TItvrm3Vz6sv-G5boJOowl8n [Oct 10 14:20:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:51] DEBUG[15619] chan_sip.c: Session timer expired: 50 - -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ [Oct 10 14:20:51] DEBUG[15619] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Oct 10 14:20:51] DEBUG[15619] chan_sip.c: ** Our prefcodec: (nothing) [Oct 10 14:20:51] DEBUG[15619] chan_sip.c: -- Done with adding codecs to SDP [Oct 10 14:20:51] DEBUG[15619] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Oct 10 14:20:51] DEBUG[15619] chan_sip.c: Initializing already initialized SIP dialog -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ (presumably reinvite) [Oct 10 14:20:51] DEBUG[15619] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:20:51] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 48 bytes [Oct 10 14:20:51] DEBUG[15619][C-00000002] logger.c: CALL_ID [C-00000002] bound to thread. [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Acked pending invite 102 [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Stopping retransmission on '-gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ' of Request 102: Match Found [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: SIP response 200 to RE-invite on outgoing call -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP o=- 76803241 76803242 IN IP4 10.24.18.165... OK. [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.165... OK. [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Oct 10 14:20:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0xb2d91ca8 [Oct 10 14:20:51] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0xb2d91ca8 [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4013 IN IP4 10.24.18.165... UNSUPPORTED OR FAILED. [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Oct 10 14:20:51] DEBUG[15619][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x912892c' [Oct 10 14:20:51] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 0 from 0xb2d91ca8 to 0x9128a54 [Oct 10 14:20:51] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 96 from 0xb2d91ca8 to 0x9128a54 [Oct 10 14:20:51] DEBUG[15619][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x912892c' [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Updating call counter for incoming call [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Session-Expires: 1800 [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Refresher: UAC [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Session timer stopped: 50 - -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Session timer started: 91 - -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ 900000ms [Oct 10 14:20:51] DEBUG[15619][C-00000002] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:20:51] DEBUG[15619][C-00000002] logger.c: CALL_ID [C-00000002] being removed from thread. [Oct 10 14:20:51] DEBUG[15890][C-00000002] res_rtp_asterisk.c: 0x912dec0 -- Probation learning mode pass with source address 10.24.18.165:4012 [Oct 10 14:20:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:20:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:20:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:20:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:04] DEBUG[16008] http.c: HTTP Request URI is /ari/bridges?&api_key=admin:secret [Oct 10 14:21:04] DEBUG[16008] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:21:04] DEBUG[16008] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:21:04] DEBUG[16008] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:21:04] DEBUG[16008] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:21:04] DEBUG[16008] http.c: Match made with [ari] [Oct 10 14:21:04] DEBUG[16008] res_ari.c: Finding handler for bridges [Oct 10 14:21:04] DEBUG[16008] res_ari.c: Checking endpoints [Oct 10 14:21:04] DEBUG[16008] res_ari.c: Checking channels [Oct 10 14:21:04] DEBUG[16008] res_ari.c: Checking events [Oct 10 14:21:04] DEBUG[16008] res_ari.c: Checking recordings [Oct 10 14:21:04] DEBUG[16008] res_ari.c: Checking playback [Oct 10 14:21:04] DEBUG[16008] res_ari.c: Checking applications [Oct 10 14:21:04] DEBUG[16008] res_ari.c: Checking bridges [Oct 10 14:21:04] DEBUG[16008] res_ari.c: Got it! [Oct 10 14:21:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:08] DEBUG[16011] http.c: HTTP Request URI is /ari/bridges?&api_key=admin:secret [Oct 10 14:21:08] DEBUG[16011] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:21:08] DEBUG[16011] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:21:08] DEBUG[16011] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:21:08] DEBUG[16011] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:21:08] DEBUG[16011] http.c: Match made with [ari] [Oct 10 14:21:08] DEBUG[16011] res_ari.c: Finding handler for bridges [Oct 10 14:21:08] DEBUG[16011] res_ari.c: Checking endpoints [Oct 10 14:21:08] DEBUG[16011] res_ari.c: Checking channels [Oct 10 14:21:08] DEBUG[16011] res_ari.c: Checking events [Oct 10 14:21:08] DEBUG[16011] res_ari.c: Checking recordings [Oct 10 14:21:08] DEBUG[16011] res_ari.c: Checking playback [Oct 10 14:21:08] DEBUG[16011] res_ari.c: Checking applications [Oct 10 14:21:08] DEBUG[16011] res_ari.c: Checking bridges [Oct 10 14:21:08] DEBUG[16011] res_ari.c: Got it! [Oct 10 14:21:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:21:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:21:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:38] DEBUG[16013] http.c: HTTP Request URI is /ari/bridges?&api_key=admin:secret [Oct 10 14:21:38] DEBUG[16013] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:21:38] DEBUG[16013] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:21:38] DEBUG[16013] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:21:38] DEBUG[16013] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:21:38] DEBUG[16013] http.c: Match made with [ari] [Oct 10 14:21:38] DEBUG[16013] res_ari.c: Finding handler for bridges [Oct 10 14:21:38] DEBUG[16013] res_ari.c: Checking endpoints [Oct 10 14:21:38] DEBUG[16013] res_ari.c: Checking channels [Oct 10 14:21:38] DEBUG[16013] res_ari.c: Checking events [Oct 10 14:21:38] DEBUG[16013] res_ari.c: Checking recordings [Oct 10 14:21:38] DEBUG[16013] res_ari.c: Checking playback [Oct 10 14:21:38] DEBUG[16013] res_ari.c: Checking applications [Oct 10 14:21:38] DEBUG[16013] res_ari.c: Checking bridges [Oct 10 14:21:38] DEBUG[16013] res_ari.c: Got it! [Oct 10 14:21:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:48] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:21:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:48] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:22:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:22:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:22:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:22:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:22:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:22:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:22:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:22:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 9KiYT2qQff84cbi5xIonS9WagxcAOhSm - SUBSCRIBE (No RTP) [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:22:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:22:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog 9KiYT2qQff84cbi5xIonS9WagxcAOhSm [Oct 10 14:22:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:22:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:15] DEBUG[16025] http.c: HTTP Request URI is /ari/bridges?&api_key=admin:secret; [Oct 10 14:23:15] DEBUG[16025] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:23:15] DEBUG[16025] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:23:15] DEBUG[16025] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:23:15] DEBUG[16025] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:23:15] DEBUG[16025] http.c: Match made with [ari] [Oct 10 14:23:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:23:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:23:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:23:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:23:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:23:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:23:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:23:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:23:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for pfQj0kGlDAbEiukjDhas91CDXKqXF8yU - SUBSCRIBE (No RTP) [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:23:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:23:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog pfQj0kGlDAbEiukjDhas91CDXKqXF8yU [Oct 10 14:23:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:23:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:23:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:48] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:23:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:23:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:23:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:23:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for YrH9rdBYchImbXbJyoHhg4x0vBxyy4tX - SUBSCRIBE (No RTP) [Oct 10 14:23:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:23:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:23:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:23:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:23:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog YrH9rdBYchImbXbJyoHhg4x0vBxyy4tX [Oct 10 14:23:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:23:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:10] DEBUG[16031] http.c: HTTP Request URI is /ari/bridges?&api_key=admin:secret [Oct 10 14:24:10] DEBUG[16031] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:24:10] DEBUG[16031] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:24:10] DEBUG[16031] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:24:10] DEBUG[16031] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:24:10] DEBUG[16031] http.c: Match made with [ari] [Oct 10 14:24:10] DEBUG[16031] res_ari.c: Finding handler for bridges [Oct 10 14:24:10] DEBUG[16031] res_ari.c: Checking endpoints [Oct 10 14:24:10] DEBUG[16031] res_ari.c: Checking channels [Oct 10 14:24:10] DEBUG[16031] res_ari.c: Checking events [Oct 10 14:24:10] DEBUG[16031] res_ari.c: Checking recordings [Oct 10 14:24:10] DEBUG[16031] res_ari.c: Checking playback [Oct 10 14:24:10] DEBUG[16031] res_ari.c: Checking applications [Oct 10 14:24:10] DEBUG[16031] res_ari.c: Checking bridges [Oct 10 14:24:10] DEBUG[16031] res_ari.c: Got it! [Oct 10 14:24:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:48] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:24:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:24:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for vDkLvZjQwVrBubP3LuLqkdryYQ4.EvAg - SUBSCRIBE (No RTP) [Oct 10 14:24:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:24:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:24:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:24:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:24:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog vDkLvZjQwVrBubP3LuLqkdryYQ4.EvAg [Oct 10 14:24:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:24:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:18] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:23] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:25:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:25:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:25:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:25:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:25:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:25:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:25:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:25:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for TKcSNDJC5LVfM8tbNBhv0w8Un1nquta8 - SUBSCRIBE (No RTP) [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:25:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:25:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog TKcSNDJC5LVfM8tbNBhv0w8Un1nquta8 [Oct 10 14:25:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:28] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:33] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:38] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:43] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:25:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:25:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:25:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:25:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:25:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:25:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:25:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:25:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for L3pdeEKmD4PI2qJQzfVzZHe3kM1DoR4b - SUBSCRIBE (No RTP) [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:25:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:25:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog L3pdeEKmD4PI2qJQzfVzZHe3kM1DoR4b [Oct 10 14:25:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:25:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:25:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:25:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:26:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:26:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:56] DEBUG[16046] http.c: HTTP Request URI is /ari/bridges? [Oct 10 14:26:56] DEBUG[16046] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:26:56] DEBUG[16046] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:26:56] DEBUG[16046] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:26:56] DEBUG[16046] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:26:56] DEBUG[16046] http.c: Match made with [ari] [Oct 10 14:26:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:26:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:16] DEBUG[16050] http.c: HTTP Request URI is /ari/bridges?&api_key=admin:secret [Oct 10 14:27:16] DEBUG[16050] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:27:16] DEBUG[16050] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:27:16] DEBUG[16050] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:27:16] DEBUG[16050] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:27:16] DEBUG[16050] http.c: Match made with [ari] [Oct 10 14:27:16] DEBUG[16050] res_ari.c: Finding handler for bridges [Oct 10 14:27:16] DEBUG[16050] res_ari.c: Checking endpoints [Oct 10 14:27:16] DEBUG[16050] res_ari.c: Checking channels [Oct 10 14:27:16] DEBUG[16050] res_ari.c: Checking events [Oct 10 14:27:16] DEBUG[16050] res_ari.c: Checking recordings [Oct 10 14:27:16] DEBUG[16050] res_ari.c: Checking playback [Oct 10 14:27:16] DEBUG[16050] res_ari.c: Checking applications [Oct 10 14:27:16] DEBUG[16050] res_ari.c: Checking bridges [Oct 10 14:27:16] DEBUG[16050] res_ari.c: Got it! [Oct 10 14:27:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:27:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:27:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:27:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:27:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:27:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:27:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:27:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:27:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for f.wwQCU7h4grT7l3EVvNWk9cHbf3UAr- - SUBSCRIBE (No RTP) [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:27:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:27:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog f.wwQCU7h4grT7l3EVvNWk9cHbf3UAr- [Oct 10 14:27:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:27:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:28:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:28:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:28:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:28:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:28:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:28:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:28:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:28:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 9YB2y7EHfmib2w02ZGZBgKGvHT1BZTcA - SUBSCRIBE (No RTP) [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:28:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:28:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog 9YB2y7EHfmib2w02ZGZBgKGvHT1BZTcA [Oct 10 14:28:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:28:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:28:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:28:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:28:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:53] DEBUG[16056] http.c: HTTP Request URI is /ari/channels?&api_key=admin:secret [Oct 10 14:28:53] DEBUG[16056] http.c: match request [ari/channels] with handler [httpstatus] len 10 [Oct 10 14:28:53] DEBUG[16056] http.c: match request [ari/channels] with handler [phoneprov] len 9 [Oct 10 14:28:53] DEBUG[16056] http.c: match request [ari/channels] with handler [static] len 6 [Oct 10 14:28:53] DEBUG[16056] http.c: match request [ari/channels] with handler [ari] len 3 [Oct 10 14:28:53] DEBUG[16056] http.c: Match made with [ari] [Oct 10 14:28:53] DEBUG[16056] res_ari.c: Finding handler for channels [Oct 10 14:28:53] DEBUG[16056] res_ari.c: Checking endpoints [Oct 10 14:28:53] DEBUG[16056] res_ari.c: Checking channels [Oct 10 14:28:53] DEBUG[16056] res_ari.c: Got it! [Oct 10 14:28:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:28:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:28:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for lRh3JKkQ0Imaqdf0X7sm8RXbvbm2vc2Y - SUBSCRIBE (No RTP) [Oct 10 14:28:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:28:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:28:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:28:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:28:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog lRh3JKkQ0Imaqdf0X7sm8RXbvbm2vc2Y [Oct 10 14:28:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:28:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:29:17] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:29:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:29:17] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:29:17] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:29:17] DEBUG[15619] chan_sip.c: Acked pending invite 105 [Oct 10 14:29:17] DEBUG[15619] chan_sip.c: Stopping retransmission on 'PLockzot5wXYYBQ89eWd18qPrTFlxMPl' of Request 105: Match Found [Oct 10 14:29:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:39] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:29:39] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:29:39] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:29:39] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:29:39] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:29:39] DEBUG[15619] chan_sip.c: Acked pending invite 105 [Oct 10 14:29:39] DEBUG[15619] chan_sip.c: Stopping retransmission on 'EVj1DqKxYCA3vR-L.ceVXGwe5iq0.Ixi' of Request 105: Match Found [Oct 10 14:29:42] DEBUG[16059] http.c: HTTP Request URI is /ari/bridges?api_key=admin%3Asecret [Oct 10 14:29:42] DEBUG[16059] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:29:42] DEBUG[16059] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:29:42] DEBUG[16059] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:29:42] DEBUG[16059] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:29:42] DEBUG[16059] http.c: Match made with [ari] [Oct 10 14:29:42] DEBUG[16059] res_ari.c: Finding handler for bridges [Oct 10 14:29:42] DEBUG[16059] res_ari.c: Checking endpoints [Oct 10 14:29:42] DEBUG[16059] res_ari.c: Checking channels [Oct 10 14:29:42] DEBUG[16059] res_ari.c: Checking events [Oct 10 14:29:42] DEBUG[16059] res_ari.c: Checking recordings [Oct 10 14:29:42] DEBUG[16059] res_ari.c: Checking playback [Oct 10 14:29:42] DEBUG[16059] res_ari.c: Checking applications [Oct 10 14:29:42] DEBUG[16059] res_ari.c: Checking bridges [Oct 10 14:29:42] DEBUG[16059] res_ari.c: Got it! [Oct 10 14:29:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:29:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:29:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for z6Hn2096T9HIUNmuIBk9nF5i2BzZi8yf - SUBSCRIBE (No RTP) [Oct 10 14:29:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:29:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:29:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:29:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:29:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog z6Hn2096T9HIUNmuIBk9nF5i2BzZi8yf [Oct 10 14:29:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:29:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:09] DEBUG[16064] http.c: HTTP Request URI is /ari/bridges?&api_key=admin:secret [Oct 10 14:30:09] DEBUG[16064] http.c: match request [ari/bridges] with handler [httpstatus] len 10 [Oct 10 14:30:09] DEBUG[16064] http.c: match request [ari/bridges] with handler [phoneprov] len 9 [Oct 10 14:30:09] DEBUG[16064] http.c: match request [ari/bridges] with handler [static] len 6 [Oct 10 14:30:09] DEBUG[16064] http.c: match request [ari/bridges] with handler [ari] len 3 [Oct 10 14:30:09] DEBUG[16064] http.c: Match made with [ari] [Oct 10 14:30:09] DEBUG[16064] res_ari.c: Finding handler for bridges [Oct 10 14:30:09] DEBUG[16064] res_ari.c: Checking endpoints [Oct 10 14:30:09] DEBUG[16064] res_ari.c: Checking channels [Oct 10 14:30:09] DEBUG[16064] res_ari.c: Checking events [Oct 10 14:30:09] DEBUG[16064] res_ari.c: Checking recordings [Oct 10 14:30:09] DEBUG[16064] res_ari.c: Checking playback [Oct 10 14:30:09] DEBUG[16064] res_ari.c: Checking applications [Oct 10 14:30:09] DEBUG[16064] res_ari.c: Checking bridges [Oct 10 14:30:09] DEBUG[16064] res_ari.c: Got it! [Oct 10 14:30:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:30:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:30:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:30:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:30:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:30:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:30:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:30:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:30:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for W.CjnrTCjGLKmGwEXZkLOVFBHQiVUxCA - SUBSCRIBE (No RTP) [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:30:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:30:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog W.CjnrTCjGLKmGwEXZkLOVFBHQiVUxCA [Oct 10 14:30:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:30:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:30:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:30:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:30:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:30:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:30:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:30:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:30:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for jhQUtQ7TQussuFb4.OaIC-83VUM7myRN - SUBSCRIBE (No RTP) [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:30:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:30:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog jhQUtQ7TQussuFb4.OaIC-83VUM7myRN [Oct 10 14:30:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:30:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:30:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:30:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:31:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:31:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:31:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:39] DEBUG[16072] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel?&api_key=admin:secret [Oct 10 14:32:39] DEBUG[16072] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [httpstatus] len 10 [Oct 10 14:32:39] DEBUG[16072] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [phoneprov] len 9 [Oct 10 14:32:39] DEBUG[16072] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [static] len 6 [Oct 10 14:32:39] DEBUG[16072] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [ari] len 3 [Oct 10 14:32:39] DEBUG[16072] http.c: Match made with [ari] [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Finding handler for bridges [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Checking endpoints [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Checking channels [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Checking events [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Checking recordings [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Checking playback [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Checking applications [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Checking bridges [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Got it! [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Checking bridgeId [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Got it! [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Finding handler for addChannel [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Checking addChannel [Oct 10 14:32:39] DEBUG[16072] res_ari.c: Got it! [Oct 10 14:32:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:32:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:32:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:32:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:32:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:32:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:32:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:32:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:32:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for xvHa5f.dbE18Gjlp8TDNUULEVJkLuFW0 - SUBSCRIBE (No RTP) [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:32:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:32:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog xvHa5f.dbE18Gjlp8TDNUULEVJkLuFW0 [Oct 10 14:32:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:32:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:33:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:33:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:33:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:33:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:33:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:33:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:33:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:33:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for bDkMd-8PXXS8CBR59qN5-BVWZDR8c9Dy - SUBSCRIBE (No RTP) [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:33:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:33:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog bDkMd-8PXXS8CBR59qN5-BVWZDR8c9Dy [Oct 10 14:33:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:33:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:33:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:33:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:33:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:33:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:33:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for nwRDyEUtpwV2s2IuMD.Vs4EPyEXgUC8K - SUBSCRIBE (No RTP) [Oct 10 14:33:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:33:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:33:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:33:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:33:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog nwRDyEUtpwV2s2IuMD.Vs4EPyEXgUC8K [Oct 10 14:33:57] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:33:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:02] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:07] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:12] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:13] DEBUG[16080] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel?&api_key=admin:secret [Oct 10 14:34:13] DEBUG[16080] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [httpstatus] len 10 [Oct 10 14:34:13] DEBUG[16080] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [phoneprov] len 9 [Oct 10 14:34:13] DEBUG[16080] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [static] len 6 [Oct 10 14:34:13] DEBUG[16080] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [ari] len 3 [Oct 10 14:34:13] DEBUG[16080] http.c: Match made with [ari] [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Finding handler for bridges [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Checking endpoints [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Checking channels [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Checking events [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Checking recordings [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Checking playback [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Checking applications [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Checking bridges [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Got it! [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Checking bridgeId [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Got it! [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Finding handler for addChannel [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Checking addChannel [Oct 10 14:34:13] DEBUG[16080] res_ari.c: Got it! [Oct 10 14:34:17] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:22] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:27] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:32] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:37] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:42] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:45] DEBUG[16082] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel?api_key=admin:secret [Oct 10 14:34:45] DEBUG[16082] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [httpstatus] len 10 [Oct 10 14:34:45] DEBUG[16082] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [phoneprov] len 9 [Oct 10 14:34:45] DEBUG[16082] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [static] len 6 [Oct 10 14:34:45] DEBUG[16082] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [ari] len 3 [Oct 10 14:34:45] DEBUG[16082] http.c: Match made with [ari] [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Finding handler for bridges [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Checking endpoints [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Checking channels [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Checking events [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Checking recordings [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Checking playback [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Checking applications [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Checking bridges [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Got it! [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Checking bridgeId [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Got it! [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Finding handler for addChannel [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Checking addChannel [Oct 10 14:34:45] DEBUG[16082] res_ari.c: Got it! [Oct 10 14:34:47] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:34:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:34:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for pA9oVXtd1nOg5WmMQT2M7scq-QzRT52v - SUBSCRIBE (No RTP) [Oct 10 14:34:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:34:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:34:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:34:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:34:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog pA9oVXtd1nOg5WmMQT2M7scq-QzRT52v [Oct 10 14:34:52] DEBUG[15888][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:53] DEBUG[16085] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel?channel=1381431943.2&api_key=admin:secret [Oct 10 14:34:53] DEBUG[16085] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [httpstatus] len 10 [Oct 10 14:34:53] DEBUG[16085] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [phoneprov] len 9 [Oct 10 14:34:53] DEBUG[16085] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [static] len 6 [Oct 10 14:34:53] DEBUG[16085] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [ari] len 3 [Oct 10 14:34:53] DEBUG[16085] http.c: Match made with [ari] [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Finding handler for bridges [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Checking endpoints [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Checking channels [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Checking events [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Checking recordings [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Checking playback [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Checking applications [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Checking bridges [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Got it! [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Checking bridgeId [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Got it! [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Finding handler for addChannel [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Checking addChannel [Oct 10 14:34:53] DEBUG[16085] res_ari.c: Got it! [Oct 10 14:34:53] DEBUG[16085] bridge_roles.c: Roles did not exist on channel SIP/phone_A-00000001 [Oct 10 14:34:53] DEBUG[16085] stasis/control.c: 1381431943.2: Sending channel add_to_bridge command [Oct 10 14:34:53] DEBUG[15888][C-00000001] stasis/control.c: 1381431943.2: Adding to bridge f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:34:53] DEBUG[16086][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x9245d9c(SIP/phone_A-00000001) [Oct 10 14:34:53] VERBOSE[16086][C-00000001] bridge_channel.c: -- Channel SIP/phone_A-00000001 joined 'simple_bridge' base-bridge [Oct 10 14:34:53] DEBUG[16086][C-00000001] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Must have two channels. [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge_native_rtp.c: Bridge 'f7c9433b-711e-4fdf-990e-7ea806eed848' can not use native RTP bridge as two channels are required [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge. [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge.c: Chose bridge technology simple_bridge [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format ulaw [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format ulaw [Oct 10 14:34:53] DEBUG[16086][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining simple_bridge technology [Oct 10 14:34:53] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 10 14:34:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:34:57] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:02] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:07] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:12] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:17] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:22] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:35:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:35:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:35:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:35:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:35:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:35:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:35:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:35:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 47riYw3-ocPStfxvKH.TgmLmYxJnb2dL - SUBSCRIBE (No RTP) [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:35:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:35:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog 47riYw3-ocPStfxvKH.TgmLmYxJnb2dL [Oct 10 14:35:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:27] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:32] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:37] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:42] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:43] DEBUG[15619] chan_sip.c: Session timer expired: 87 - iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX [Oct 10 14:35:43] DEBUG[15619] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Oct 10 14:35:43] DEBUG[15619] chan_sip.c: ** Our prefcodec: (nothing) [Oct 10 14:35:43] DEBUG[15619] chan_sip.c: -- Done with adding codecs to SDP [Oct 10 14:35:43] DEBUG[15619] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Oct 10 14:35:43] DEBUG[15619] chan_sip.c: Initializing already initialized SIP dialog iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX (presumably reinvite) [Oct 10 14:35:43] DEBUG[15619] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:35:43] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 48 bytes [Oct 10 14:35:44] DEBUG[15619][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Acked pending invite 103 [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Stopping retransmission on 'iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX' of Request 103: Match Found [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: SIP response 200 to RE-invite on outgoing call iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP o=- 76803233 76803235 IN IP4 10.24.19.97... OK. [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.19.97... OK. [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Oct 10 14:35:44] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0xb2d91ca8 [Oct 10 14:35:44] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0xb2d91ca8 [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4031 IN IP4 10.24.19.97... UNSUPPORTED OR FAILED. [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Oct 10 14:35:44] DEBUG[15619][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb2a0a374' [Oct 10 14:35:44] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 0 from 0xb2d91ca8 to 0xb2a0a49c [Oct 10 14:35:44] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 96 from 0xb2d91ca8 to 0xb2a0a49c [Oct 10 14:35:44] DEBUG[15619][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb2a0a374' [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: We have an owner, now see if we need to change this call [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Updating call counter for incoming call [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Session-Expires: 1800 [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Refresher: UAC [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Session timer stopped: 87 - iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Session timer started: 119 - iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX 900000ms [Oct 10 14:35:44] DEBUG[15619][C-00000001] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:35:44] DEBUG[15619][C-00000001] logger.c: CALL_ID [C-00000001] being removed from thread. [Oct 10 14:35:44] DEBUG[16086][C-00000001] res_rtp_asterisk.c: 0xb2cc9aa8 -- Probation learning mode pass with source address 10.24.19.97:4030 [Oct 10 14:35:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:35:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:35:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:35:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:35:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:35:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:35:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:35:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:35:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Z4DZ2ZBHUhHY88dj60r23vscowJ.Z9om - SUBSCRIBE (No RTP) [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:35:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:35:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog Z4DZ2ZBHUhHY88dj60r23vscowJ.Z9om [Oct 10 14:35:47] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:51] DEBUG[15619] chan_sip.c: Session timer expired: 91 - -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ [Oct 10 14:35:51] DEBUG[15619] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Oct 10 14:35:51] DEBUG[15619] chan_sip.c: ** Our prefcodec: (nothing) [Oct 10 14:35:51] DEBUG[15619] chan_sip.c: -- Done with adding codecs to SDP [Oct 10 14:35:51] DEBUG[15619] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Oct 10 14:35:51] DEBUG[15619] chan_sip.c: Initializing already initialized SIP dialog -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ (presumably reinvite) [Oct 10 14:35:51] DEBUG[15619] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:35:51] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 48 bytes [Oct 10 14:35:52] DEBUG[15619][C-00000002] logger.c: CALL_ID [C-00000002] bound to thread. [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Acked pending invite 103 [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Stopping retransmission on '-gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ' of Request 103: Match Found [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: SIP response 200 to RE-invite on outgoing call -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP o=- 76803241 76803243 IN IP4 10.24.18.165... OK. [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.165... OK. [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Oct 10 14:35:52] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0xb2d91ca8 [Oct 10 14:35:52] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0xb2d91ca8 [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4013 IN IP4 10.24.18.165... UNSUPPORTED OR FAILED. [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Oct 10 14:35:52] DEBUG[15619][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x912892c' [Oct 10 14:35:52] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 0 from 0xb2d91ca8 to 0x9128a54 [Oct 10 14:35:52] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 96 from 0xb2d91ca8 to 0x9128a54 [Oct 10 14:35:52] DEBUG[15619][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x912892c' [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Updating call counter for incoming call [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Session-Expires: 1800 [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Refresher: UAC [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Session timer stopped: 91 - -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Session timer started: 123 - -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ 900000ms [Oct 10 14:35:52] DEBUG[15619][C-00000002] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:35:52] DEBUG[15619][C-00000002] logger.c: CALL_ID [C-00000002] being removed from thread. [Oct 10 14:35:52] DEBUG[15890][C-00000002] res_rtp_asterisk.c: 0x912dec0 -- Probation learning mode pass with source address 10.24.18.165:4012 [Oct 10 14:35:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:53] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:35:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:35:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:35:58] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:36:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:36:03] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:36:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:36:08] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:36:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:36:13] DEBUG[15890][C-00000002] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:36:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 40 bytes [Oct 10 14:36:17] DEBUG[16098] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel?channel=1381431951.3&api_key=admin:secret [Oct 10 14:36:17] DEBUG[16098] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [httpstatus] len 10 [Oct 10 14:36:17] DEBUG[16098] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [phoneprov] len 9 [Oct 10 14:36:17] DEBUG[16098] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [static] len 6 [Oct 10 14:36:17] DEBUG[16098] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/addChannel] with handler [ari] len 3 [Oct 10 14:36:17] DEBUG[16098] http.c: Match made with [ari] [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Finding handler for bridges [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Checking endpoints [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Checking channels [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Checking events [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Checking recordings [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Checking playback [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Checking applications [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Checking bridges [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Got it! [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Checking bridgeId [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Got it! [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Finding handler for addChannel [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Checking addChannel [Oct 10 14:36:17] DEBUG[16098] res_ari.c: Got it! [Oct 10 14:36:17] DEBUG[16098] bridge_roles.c: Roles did not exist on channel SIP/phone_B-00000002 [Oct 10 14:36:17] DEBUG[16098] stasis/control.c: 1381431951.3: Sending channel add_to_bridge command [Oct 10 14:36:17] DEBUG[15890][C-00000002] stasis/control.c: 1381431951.3: Adding to bridge f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:36:17] DEBUG[16099][C-00000002] logger.c: CALL_ID [C-00000002] bound to thread. [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x9248004(SIP/phone_B-00000002) [Oct 10 14:36:17] VERBOSE[16099][C-00000002] bridge_channel.c: -- Channel SIP/phone_B-00000002 joined 'simple_bridge' base-bridge [Oct 10 14:36:17] DEBUG[15568] cdr.c: Finalized CDR for SIP/phone_B-00000002 - start 1381431951.405192 answer 1381431951.406167 end 1381433777.693711 dispo ANSWERED [Oct 10 14:36:17] DEBUG[16099][C-00000002] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:36:17] DEBUG[16099][C-00000002] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Chose bridge technology native_rtp [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling simple_bridge technology stop [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving simple_bridge technology (dummy) [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format ulaw [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format ulaw [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format ulaw [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format ulaw [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:36:17] DEBUG[16099][C-00000002] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling simple_bridge technology destructor [Oct 10 14:36:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 10 14:36:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Oct 10 14:36:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Oct 10 14:36:18] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:36:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:36:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:23] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:28] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:33] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:38] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:42] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:36:42] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.17.17:5060 [Oct 10 14:36:42] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.17.17:5060 [Oct 10 14:36:42] DEBUG[15619] chan_sip.c: Acked pending invite 103 [Oct 10 14:36:42] DEBUG[15619] chan_sip.c: Stopping retransmission on '5256f79b557e-q8enuxwifjh3' of Request 103: Match Found [Oct 10 14:36:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:43] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:48] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:53] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:36:58] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:03] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:08] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:13] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:18] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:23] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:28] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:33] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:38] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:43] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:48] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:53] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:37:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:37:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:37:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:37:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:37:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:37:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:37:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:37:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 9PIMwrSlLi-MOZhMshX1wYf199Sp2Wrb - SUBSCRIBE (No RTP) [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:37:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:37:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog 9PIMwrSlLi-MOZhMshX1wYf199Sp2Wrb [Oct 10 14:37:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:37:58] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:03] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:08] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:13] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:38:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:38:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:38:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:38:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:38:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:38:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:38:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:38:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Kyl2CTinNqmwGfi-yy5e37wOPK4cz-bl - SUBSCRIBE (No RTP) [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:38:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:38:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog Kyl2CTinNqmwGfi-yy5e37wOPK4cz-bl [Oct 10 14:38:18] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:23] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:26] DEBUG[16108] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:38:26] DEBUG[16108] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:38:26] DEBUG[16108] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:38:26] DEBUG[16108] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:38:26] DEBUG[16108] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:38:26] DEBUG[16108] http.c: Match made with [ari] [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Finding handler for bridges [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking endpoints [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking channels [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking events [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking recordings [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking playback [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking applications [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking bridges [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Got it! [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking bridgeId [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Got it! [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Finding handler for play [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking addChannel [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking removeChannel [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking mohStart [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking mohStop [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Checking play [Oct 10 14:38:26] DEBUG[16108] res_ari.c: Got it! [Oct 10 14:38:26] DEBUG[16108] bridge_roles.c: Set role 'announcer' [Oct 10 14:38:26] DEBUG[16108] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000000;1' [Oct 10 14:38:26] DEBUG[16109] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260748c(Announcer/ARI-00000000;2) is joining [Oct 10 14:38:26] DEBUG[16108] res_stasis_playback.c: 1381433906.4: Sending play(sound:demo-congrats) command [Oct 10 14:38:26] DEBUG[16109] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb260748c(Announcer/ARI-00000000;2) [Oct 10 14:38:26] DEBUG[16109] bridge_roles.c: Set role 'announcer' [Oct 10 14:38:26] VERBOSE[16109] bridge_channel.c: -- Channel Announcer/ARI-00000000;2 joined 'native_rtp' base-bridge [Oct 10 14:38:26] DEBUG[16110][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:38:26] DEBUG[16109] bridge.c: Chose bridge technology softmix [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:38:26] DEBUG[16110][C-00000001] channel.c: Set channel Announcer/ARI-00000000;1 to write format gsm [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge technology softmix wants to read any of formats (slin) but channel has ulaw [Oct 10 14:38:26] DEBUG[16109] channel.c: Set channel SIP/phone_A-00000001 to read format slin [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 put channel SIP/phone_A-00000001 into read format slin [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge technology softmix wants to write any of formats (slin) but channel has ulaw [Oct 10 14:38:26] DEBUG[16109] channel.c: Set channel SIP/phone_A-00000001 to write format slin [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 put channel SIP/phone_A-00000001 into write format slin [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:38:26] DEBUG[16109] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:38:26] DEBUG[16109] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge technology softmix wants to read any of formats (slin) but channel has ulaw [Oct 10 14:38:26] DEBUG[16109] channel.c: Set channel SIP/phone_B-00000002 to read format slin [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 put channel SIP/phone_B-00000002 into read format slin [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge technology softmix wants to write any of formats (slin) but channel has ulaw [Oct 10 14:38:26] DEBUG[16109] channel.c: Set channel SIP/phone_B-00000002 to write format slin [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 put channel SIP/phone_B-00000002 into write format slin [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:38:26] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 170488846 to 2013607224 due to a source change [Oct 10 14:38:26] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet [Oct 10 14:38:26] DEBUG[16109] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:38:26] DEBUG[16109] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000000;2 already has read format slin [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000000;2 already has write format slin [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260748c(Announcer/ARI-00000000;2) is joining softmix technology [Oct 10 14:38:26] DEBUG[16109] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:38:26] DEBUG[16109] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:38:26] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:38:26] DEBUG[16111][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:38:26] DEBUG[16111][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:38:26] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw [Oct 10 14:38:26] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 1027976, ms is 128517 [Oct 10 14:38:26] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160 [Oct 10 14:38:26] DEBUG[16110][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:38:26] VERBOSE[16110][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:38:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:38:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:38:28] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:33] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:38] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:43] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:48] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:38:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:38:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:53] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:54] DEBUG[16110][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:38:54] DEBUG[16110][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:38:54] DEBUG[16110][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:38:54] DEBUG[16110][C-00000001] channel.c: Set channel Announcer/ARI-00000000;1 to write format slin [Oct 10 14:38:54] DEBUG[16110][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000000;1' [Oct 10 14:38:54] DEBUG[16109] bridge_channel.c: Setting 0xb260748c(Announcer/ARI-00000000;2) state from:0 to:1 [Oct 10 14:38:54] DEBUG[16109] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb260748c(Announcer/ARI-00000000;2) [Oct 10 14:38:54] VERBOSE[16109] bridge_channel.c: -- Channel Announcer/ARI-00000000;2 left 'softmix' base-bridge [Oct 10 14:38:54] DEBUG[16109] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260748c(Announcer/ARI-00000000;2) is leaving softmix technology [Oct 10 14:38:54] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:38:54] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:38:54] DEBUG[16109] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:38:54] DEBUG[16109] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:38:54] DEBUG[16109] bridge.c: Chose bridge technology native_rtp [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:38:54] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1656865830 to 258596725 due to a source change [Oct 10 14:38:54] DEBUG[16109] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:38:54] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 2013607224 to 401077873 due to a source change [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:38:54] DEBUG[16109] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:38:54] DEBUG[16109] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:38:54] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:38:54] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:38:54] DEBUG[15567][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:38:54] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:38:54] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:38:54] DEBUG[16109] channel.c: Hanging up channel 'Announcer/ARI-00000000;2' [Oct 10 14:38:54] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:38:54] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:38:54] DEBUG[16111][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:38:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:38:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:38:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 5mG5g158jl4IvQbkf-m9cYISforlMSzs - SUBSCRIBE (No RTP) [Oct 10 14:38:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:38:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:38:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:38:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:38:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog 5mG5g158jl4IvQbkf-m9cYISforlMSzs [Oct 10 14:38:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:38:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:12] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:39:12] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:39:12] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:39:12] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:39:12] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:39:12] DEBUG[15619] chan_sip.c: Acked pending invite 106 [Oct 10 14:39:12] DEBUG[15619] chan_sip.c: Stopping retransmission on 'PLockzot5wXYYBQ89eWd18qPrTFlxMPl' of Request 106: Match Found [Oct 10 14:39:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:34] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:39:34] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:39:34] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:39:34] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:39:34] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:39:34] DEBUG[15619] chan_sip.c: Acked pending invite 106 [Oct 10 14:39:34] DEBUG[15619] chan_sip.c: Stopping retransmission on 'EVj1DqKxYCA3vR-L.ceVXGwe5iq0.Ixi' of Request 106: Match Found [Oct 10 14:39:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:39:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:39:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for HL5zp7gR7ISqCjiZTd5g9eIWEhOVwEG9 - SUBSCRIBE (No RTP) [Oct 10 14:39:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:39:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:39:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:39:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:39:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog HL5zp7gR7ISqCjiZTd5g9eIWEhOVwEG9 [Oct 10 14:39:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:39:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:40:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:40:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:40:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:40:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:40:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:40:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:40:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:40:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for DnQu4ipZxw4cwHpI.DyXI0ubIRBtJbk4 - SUBSCRIBE (No RTP) [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:40:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:40:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog DnQu4ipZxw4cwHpI.DyXI0ubIRBtJbk4 [Oct 10 14:40:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:28] DEBUG[16128] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:40:28] DEBUG[16128] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:40:28] DEBUG[16128] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:40:28] DEBUG[16128] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:40:28] DEBUG[16128] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:40:28] DEBUG[16128] http.c: Match made with [ari] [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Finding handler for bridges [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking endpoints [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking channels [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking events [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking recordings [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking playback [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking applications [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking bridges [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Got it! [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking bridgeId [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Got it! [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Finding handler for play [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking addChannel [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking removeChannel [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking mohStart [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking mohStop [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Checking play [Oct 10 14:40:28] DEBUG[16128] res_ari.c: Got it! [Oct 10 14:40:28] DEBUG[16128] bridge_roles.c: Set role 'announcer' [Oct 10 14:40:28] DEBUG[16128] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000001;1' [Oct 10 14:40:28] DEBUG[16128] res_stasis_playback.c: 1381434028.6: Sending play(sound:demo-congrats) command [Oct 10 14:40:28] DEBUG[16129] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2808b8c(Announcer/ARI-00000001;2) is joining [Oct 10 14:40:28] DEBUG[16130][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:40:28] DEBUG[16129] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2808b8c(Announcer/ARI-00000001;2) [Oct 10 14:40:28] DEBUG[16129] bridge_roles.c: Set role 'announcer' [Oct 10 14:40:28] VERBOSE[16129] bridge_channel.c: -- Channel Announcer/ARI-00000001;2 joined 'native_rtp' base-bridge [Oct 10 14:40:28] DEBUG[16130][C-00000001] channel.c: Set channel Announcer/ARI-00000001;1 to write format gsm [Oct 10 14:40:28] DEBUG[16130][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:40:28] VERBOSE[16130][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:40:28] DEBUG[16129] bridge.c: Chose bridge technology softmix [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:40:28] DEBUG[16129] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:40:28] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 258596725 to 1308713568 due to a source change [Oct 10 14:40:28] DEBUG[16129] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:40:28] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 401077873 to 1426476170 due to a source change [Oct 10 14:40:28] DEBUG[16129] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:40:28] DEBUG[16129] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000001;2 already has read format slin [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000001;2 already has write format slin [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2808b8c(Announcer/ARI-00000001;2) is joining softmix technology [Oct 10 14:40:28] DEBUG[16129] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:40:28] DEBUG[16129] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:40:28] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:40:28] DEBUG[16131][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:40:28] DEBUG[16131][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:40:28] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 756320, ms is 94560 [Oct 10 14:40:28] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 756320, ms is 94560 [Oct 10 14:40:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:40] DEBUG[16134] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:40:40] DEBUG[16134] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:40:40] DEBUG[16134] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:40:40] DEBUG[16134] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:40:40] DEBUG[16134] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:40:40] DEBUG[16134] http.c: Match made with [ari] [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Finding handler for bridges [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking endpoints [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking channels [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking events [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking recordings [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking playback [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking applications [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking bridges [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Got it! [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking bridgeId [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Got it! [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Finding handler for play [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking addChannel [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking removeChannel [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking mohStart [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking mohStop [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Checking play [Oct 10 14:40:40] DEBUG[16134] res_ari.c: Got it! [Oct 10 14:40:40] DEBUG[16134] bridge_roles.c: Set role 'announcer' [Oct 10 14:40:40] DEBUG[16134] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000002;1' [Oct 10 14:40:40] DEBUG[16134] res_stasis_playback.c: 1381434040.8: Sending play(sound:demo-congrats) command [Oct 10 14:40:40] DEBUG[16135] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9105dac(Announcer/ARI-00000002;2) is joining [Oct 10 14:40:40] DEBUG[16135] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x9105dac(Announcer/ARI-00000002;2) [Oct 10 14:40:40] DEBUG[16135] bridge_roles.c: Set role 'announcer' [Oct 10 14:40:40] VERBOSE[16135] bridge_channel.c: -- Channel Announcer/ARI-00000002;2 joined 'softmix' base-bridge [Oct 10 14:40:40] DEBUG[16136][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:40:40] DEBUG[16135] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:40:40] DEBUG[16135] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:40:40] DEBUG[16135] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:40:40] DEBUG[16135] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:40:40] DEBUG[16135] bridge.c: Chose bridge technology softmix [Oct 10 14:40:40] DEBUG[16135] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:40:40] DEBUG[16135] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000002;2 already has read format slin [Oct 10 14:40:40] DEBUG[16135] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000002;2 already has write format slin [Oct 10 14:40:40] DEBUG[16135] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9105dac(Announcer/ARI-00000002;2) is joining softmix technology [Oct 10 14:40:40] DEBUG[16135] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:40:40] DEBUG[16135] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:40:40] DEBUG[16136][C-00000001] channel.c: Set channel Announcer/ARI-00000002;1 to write format gsm [Oct 10 14:40:40] DEBUG[16136][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:40:40] VERBOSE[16136][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:40:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:40:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:40:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:40:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:40:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:40:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:40:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:40:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:40:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for IpYw8CF3caa3bXjET40p4YBOxk9R3dsm - SUBSCRIBE (No RTP) [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:40:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:40:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog IpYw8CF3caa3bXjET40p4YBOxk9R3dsm [Oct 10 14:40:47] DEBUG[16140] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:40:47] DEBUG[16140] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:40:47] DEBUG[16140] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:40:47] DEBUG[16140] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:40:47] DEBUG[16140] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:40:47] DEBUG[16140] http.c: Match made with [ari] [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Finding handler for bridges [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking endpoints [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking channels [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking events [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking recordings [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking playback [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking applications [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking bridges [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Got it! [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking bridgeId [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Got it! [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Finding handler for play [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking addChannel [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking removeChannel [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking mohStart [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking mohStop [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Checking play [Oct 10 14:40:47] DEBUG[16140] res_ari.c: Got it! [Oct 10 14:40:47] DEBUG[16140] bridge_roles.c: Set role 'announcer' [Oct 10 14:40:47] DEBUG[16140] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000003;1' [Oct 10 14:40:47] DEBUG[16140] res_stasis_playback.c: 1381434047.10: Sending play(sound:demo-congrats) command [Oct 10 14:40:47] DEBUG[16141] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2bf3fec(Announcer/ARI-00000003;2) is joining [Oct 10 14:40:47] DEBUG[16141] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2bf3fec(Announcer/ARI-00000003;2) [Oct 10 14:40:47] DEBUG[16141] bridge_roles.c: Set role 'announcer' [Oct 10 14:40:47] VERBOSE[16141] bridge_channel.c: -- Channel Announcer/ARI-00000003;2 joined 'softmix' base-bridge [Oct 10 14:40:47] DEBUG[16142][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:40:47] DEBUG[16141] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:40:47] DEBUG[16141] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:40:47] DEBUG[16141] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:40:47] DEBUG[16141] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:40:47] DEBUG[16141] bridge.c: Chose bridge technology softmix [Oct 10 14:40:47] DEBUG[16141] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:40:47] DEBUG[16141] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000003;2 already has read format slin [Oct 10 14:40:47] DEBUG[16141] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000003;2 already has write format slin [Oct 10 14:40:47] DEBUG[16141] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2bf3fec(Announcer/ARI-00000003;2) is joining softmix technology [Oct 10 14:40:47] DEBUG[16141] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:40:47] DEBUG[16141] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:40:47] DEBUG[16142][C-00000001] channel.c: Set channel Announcer/ARI-00000003;1 to write format gsm [Oct 10 14:40:47] DEBUG[16142][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:40:47] VERBOSE[16142][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:40:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:52] DEBUG[16145] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:40:52] DEBUG[16145] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:40:52] DEBUG[16145] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:40:52] DEBUG[16145] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:40:52] DEBUG[16145] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:40:52] DEBUG[16145] http.c: Match made with [ari] [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Finding handler for bridges [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking endpoints [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking channels [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking events [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking recordings [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking playback [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking applications [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking bridges [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Got it! [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking bridgeId [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Got it! [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Finding handler for play [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking addChannel [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking removeChannel [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking mohStart [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking mohStop [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Checking play [Oct 10 14:40:52] DEBUG[16145] res_ari.c: Got it! [Oct 10 14:40:52] DEBUG[16145] bridge_roles.c: Set role 'announcer' [Oct 10 14:40:52] DEBUG[16145] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000004;1' [Oct 10 14:40:52] DEBUG[16145] res_stasis_playback.c: 1381434052.12: Sending play(sound:demo-congrats) command [Oct 10 14:40:52] DEBUG[16146] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x920d79c(Announcer/ARI-00000004;2) is joining [Oct 10 14:40:52] DEBUG[16146] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x920d79c(Announcer/ARI-00000004;2) [Oct 10 14:40:52] DEBUG[16147][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:40:52] DEBUG[16146] bridge_roles.c: Set role 'announcer' [Oct 10 14:40:52] VERBOSE[16146] bridge_channel.c: -- Channel Announcer/ARI-00000004;2 joined 'softmix' base-bridge [Oct 10 14:40:52] DEBUG[16147][C-00000001] channel.c: Set channel Announcer/ARI-00000004;1 to write format gsm [Oct 10 14:40:52] DEBUG[16146] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:40:52] DEBUG[16146] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:40:52] DEBUG[16146] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:40:52] DEBUG[16146] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:40:52] DEBUG[16146] bridge.c: Chose bridge technology softmix [Oct 10 14:40:52] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:40:52] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000004;2 already has read format slin [Oct 10 14:40:52] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000004;2 already has write format slin [Oct 10 14:40:52] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x920d79c(Announcer/ARI-00000004;2) is joining softmix technology [Oct 10 14:40:52] DEBUG[16147][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:40:52] VERBOSE[16147][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:40:52] DEBUG[16146] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:40:52] DEBUG[16146] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:40:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:56] DEBUG[16130][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:40:56] DEBUG[16130][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:40:56] DEBUG[16130][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:40:56] DEBUG[16130][C-00000001] channel.c: Set channel Announcer/ARI-00000001;1 to write format slin [Oct 10 14:40:56] DEBUG[16130][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000001;1' [Oct 10 14:40:56] DEBUG[16129] bridge_channel.c: Setting 0xb2808b8c(Announcer/ARI-00000001;2) state from:0 to:1 [Oct 10 14:40:56] DEBUG[16129] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2808b8c(Announcer/ARI-00000001;2) [Oct 10 14:40:56] VERBOSE[16129] bridge_channel.c: -- Channel Announcer/ARI-00000001;2 left 'softmix' base-bridge [Oct 10 14:40:56] DEBUG[16129] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2808b8c(Announcer/ARI-00000001;2) is leaving softmix technology [Oct 10 14:40:56] DEBUG[16129] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:40:56] DEBUG[16129] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:40:56] DEBUG[16129] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:40:56] DEBUG[16129] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:40:56] DEBUG[16129] bridge.c: Chose bridge technology softmix [Oct 10 14:40:56] DEBUG[16129] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:40:56] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:40:56] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:40:56] DEBUG[16129] channel.c: Hanging up channel 'Announcer/ARI-00000001;2' [Oct 10 14:40:56] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:40:56] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:40:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:40:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:40:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:40:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:08] DEBUG[16136][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:41:08] DEBUG[16136][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:41:08] DEBUG[16136][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:41:08] DEBUG[16136][C-00000001] channel.c: Set channel Announcer/ARI-00000002;1 to write format slin [Oct 10 14:41:08] DEBUG[16136][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000002;1' [Oct 10 14:41:08] DEBUG[16135] bridge_channel.c: Setting 0x9105dac(Announcer/ARI-00000002;2) state from:0 to:1 [Oct 10 14:41:08] DEBUG[16135] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0x9105dac(Announcer/ARI-00000002;2) [Oct 10 14:41:08] VERBOSE[16135] bridge_channel.c: -- Channel Announcer/ARI-00000002;2 left 'softmix' base-bridge [Oct 10 14:41:08] DEBUG[16135] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9105dac(Announcer/ARI-00000002;2) is leaving softmix technology [Oct 10 14:41:08] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:41:08] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:41:08] DEBUG[16135] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:41:08] DEBUG[16135] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:41:08] DEBUG[16135] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:41:08] DEBUG[16135] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:41:08] DEBUG[16135] bridge.c: Chose bridge technology softmix [Oct 10 14:41:08] DEBUG[16135] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:41:08] DEBUG[16135] channel.c: Hanging up channel 'Announcer/ARI-00000002;2' [Oct 10 14:41:08] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:41:08] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:41:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:15] DEBUG[16142][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:41:15] DEBUG[16142][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:41:15] DEBUG[16142][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:41:15] DEBUG[16142][C-00000001] channel.c: Set channel Announcer/ARI-00000003;1 to write format slin [Oct 10 14:41:15] DEBUG[16142][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000003;1' [Oct 10 14:41:15] DEBUG[16141] bridge_channel.c: Setting 0xb2bf3fec(Announcer/ARI-00000003;2) state from:0 to:1 [Oct 10 14:41:15] DEBUG[16141] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2bf3fec(Announcer/ARI-00000003;2) [Oct 10 14:41:15] VERBOSE[16141] bridge_channel.c: -- Channel Announcer/ARI-00000003;2 left 'softmix' base-bridge [Oct 10 14:41:15] DEBUG[16141] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2bf3fec(Announcer/ARI-00000003;2) is leaving softmix technology [Oct 10 14:41:15] DEBUG[16141] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:41:15] DEBUG[16141] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:41:15] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:41:15] DEBUG[16141] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:41:15] DEBUG[16141] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:41:15] DEBUG[16141] bridge.c: Chose bridge technology softmix [Oct 10 14:41:15] DEBUG[16141] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:41:15] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:41:15] DEBUG[16141] channel.c: Hanging up channel 'Announcer/ARI-00000003;2' [Oct 10 14:41:15] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:41:15] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:41:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:41:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:41:19] DEBUG[16147][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:41:19] DEBUG[16147][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:41:19] DEBUG[16147][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:41:19] DEBUG[16147][C-00000001] channel.c: Set channel Announcer/ARI-00000004;1 to write format slin [Oct 10 14:41:19] DEBUG[16147][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000004;1' [Oct 10 14:41:19] DEBUG[16146] bridge_channel.c: Setting 0x920d79c(Announcer/ARI-00000004;2) state from:0 to:1 [Oct 10 14:41:19] DEBUG[16146] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0x920d79c(Announcer/ARI-00000004;2) [Oct 10 14:41:19] VERBOSE[16146] bridge_channel.c: -- Channel Announcer/ARI-00000004;2 left 'softmix' base-bridge [Oct 10 14:41:19] DEBUG[16146] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x920d79c(Announcer/ARI-00000004;2) is leaving softmix technology [Oct 10 14:41:19] DEBUG[16146] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:41:19] DEBUG[16146] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:41:19] DEBUG[16146] bridge.c: Chose bridge technology native_rtp [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:41:19] DEBUG[16146] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:41:19] DEBUG[16146] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:41:19] DEBUG[16146] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:41:19] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1426476170 to 1013042198 due to a source change [Oct 10 14:41:19] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1308713568 to 1543440097 due to a source change [Oct 10 14:41:19] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:41:19] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:41:19] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:41:19] DEBUG[16146] channel.c: Hanging up channel 'Announcer/ARI-00000004;2' [Oct 10 14:41:19] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:41:19] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:41:19] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:41:19] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:41:19] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:41:19] DEBUG[16131][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:41:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:41:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:25] DEBUG[16155] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:42:25] DEBUG[16155] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:42:25] DEBUG[16155] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:42:25] DEBUG[16155] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:42:25] DEBUG[16155] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:42:25] DEBUG[16155] http.c: Match made with [ari] [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Finding handler for bridges [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking endpoints [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking channels [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking events [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking recordings [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking playback [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking applications [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking bridges [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Got it! [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking bridgeId [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Got it! [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Finding handler for play [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking addChannel [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking removeChannel [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking mohStart [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking mohStop [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Checking play [Oct 10 14:42:25] DEBUG[16155] res_ari.c: Got it! [Oct 10 14:42:25] DEBUG[16155] bridge_roles.c: Set role 'announcer' [Oct 10 14:42:25] DEBUG[16155] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000005;1' [Oct 10 14:42:25] DEBUG[16155] res_stasis_playback.c: 1381434145.14: Sending play(sound:demo-congrats) command [Oct 10 14:42:25] DEBUG[16156] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2606b24(Announcer/ARI-00000005;2) is joining [Oct 10 14:42:25] DEBUG[16156] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2606b24(Announcer/ARI-00000005;2) [Oct 10 14:42:25] DEBUG[16156] bridge_roles.c: Set role 'announcer' [Oct 10 14:42:25] VERBOSE[16156] bridge_channel.c: -- Channel Announcer/ARI-00000005;2 joined 'native_rtp' base-bridge [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:42:25] DEBUG[16156] bridge.c: Chose bridge technology softmix [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:42:25] DEBUG[16157][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:42:25] DEBUG[16157][C-00000001] channel.c: Set channel Announcer/ARI-00000005;1 to write format gsm [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:42:25] DEBUG[16156] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:42:25] DEBUG[16156] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:42:25] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1013042198 to 2043565869 due to a source change [Oct 10 14:42:25] DEBUG[16157][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:42:25] DEBUG[16156] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:42:25] DEBUG[16156] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:42:25] VERBOSE[16157][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000005;2 already has read format slin [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000005;2 already has write format slin [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2606b24(Announcer/ARI-00000005;2) is joining softmix technology [Oct 10 14:42:25] DEBUG[16156] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:42:25] DEBUG[16156] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:42:25] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:42:25] DEBUG[16158][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:42:25] DEBUG[16158][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:42:25] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1543440097 to 876373701 due to a source change [Oct 10 14:42:25] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 527448, ms is 65951 [Oct 10 14:42:25] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 527448, ms is 65951 [Oct 10 14:42:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:50] DEBUG[16162] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:42:50] DEBUG[16162] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:42:50] DEBUG[16162] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:42:50] DEBUG[16162] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:42:50] DEBUG[16162] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:42:50] DEBUG[16162] http.c: Match made with [ari] [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Finding handler for bridges [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking endpoints [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking channels [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking events [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking recordings [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking playback [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking applications [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking bridges [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Got it! [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking bridgeId [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Got it! [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Finding handler for play [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking addChannel [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking removeChannel [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking mohStart [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking mohStop [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Checking play [Oct 10 14:42:50] DEBUG[16162] res_ari.c: Got it! [Oct 10 14:42:50] DEBUG[16162] bridge_roles.c: Set role 'announcer' [Oct 10 14:42:50] DEBUG[16162] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000006;1' [Oct 10 14:42:50] DEBUG[16162] res_stasis_playback.c: 1381434170.16: Sending play(sound:demo-congrats) command [Oct 10 14:42:50] DEBUG[16163] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a892ac(Announcer/ARI-00000006;2) is joining [Oct 10 14:42:50] DEBUG[16164][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:42:50] DEBUG[16163] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a892ac(Announcer/ARI-00000006;2) [Oct 10 14:42:50] DEBUG[16163] bridge_roles.c: Set role 'announcer' [Oct 10 14:42:50] VERBOSE[16163] bridge_channel.c: -- Channel Announcer/ARI-00000006;2 joined 'softmix' base-bridge [Oct 10 14:42:50] DEBUG[16164][C-00000001] channel.c: Set channel Announcer/ARI-00000006;1 to write format gsm [Oct 10 14:42:50] DEBUG[16163] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:42:50] DEBUG[16163] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:42:50] DEBUG[16163] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:42:50] DEBUG[16163] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:42:50] DEBUG[16163] bridge.c: Chose bridge technology softmix [Oct 10 14:42:50] DEBUG[16163] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:42:50] DEBUG[16163] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000006;2 already has read format slin [Oct 10 14:42:50] DEBUG[16163] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000006;2 already has write format slin [Oct 10 14:42:50] DEBUG[16163] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a892ac(Announcer/ARI-00000006;2) is joining softmix technology [Oct 10 14:42:50] DEBUG[16163] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:42:50] DEBUG[16163] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:42:50] DEBUG[16164][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:42:50] VERBOSE[16164][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:42:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:53] DEBUG[16157][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:42:53] DEBUG[16157][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:42:53] DEBUG[16157][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:42:53] DEBUG[16157][C-00000001] channel.c: Set channel Announcer/ARI-00000005;1 to write format slin [Oct 10 14:42:53] DEBUG[16157][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000005;1' [Oct 10 14:42:53] DEBUG[16156] bridge_channel.c: Setting 0xb2606b24(Announcer/ARI-00000005;2) state from:0 to:1 [Oct 10 14:42:53] DEBUG[16156] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2606b24(Announcer/ARI-00000005;2) [Oct 10 14:42:53] VERBOSE[16156] bridge_channel.c: -- Channel Announcer/ARI-00000005;2 left 'softmix' base-bridge [Oct 10 14:42:53] DEBUG[16156] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2606b24(Announcer/ARI-00000005;2) is leaving softmix technology [Oct 10 14:42:53] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:42:53] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:42:53] DEBUG[16156] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:42:53] DEBUG[16156] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:42:53] DEBUG[16156] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:42:53] DEBUG[16156] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:42:53] DEBUG[16156] bridge.c: Chose bridge technology softmix [Oct 10 14:42:53] DEBUG[16156] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:42:53] DEBUG[16156] channel.c: Hanging up channel 'Announcer/ARI-00000005;2' [Oct 10 14:42:53] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:42:53] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:42:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:42:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:42:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:42:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:42:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:42:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:42:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:42:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:42:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for YH6VuO-8XDd4X3HZ29ynpUFFoZP-Tgw. - SUBSCRIBE (No RTP) [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:42:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:42:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog YH6VuO-8XDd4X3HZ29ynpUFFoZP-Tgw. [Oct 10 14:42:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:42:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:08] DEBUG[16169] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:43:08] DEBUG[16169] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:43:08] DEBUG[16169] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:43:08] DEBUG[16169] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:43:08] DEBUG[16169] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:43:08] DEBUG[16169] http.c: Match made with [ari] [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Finding handler for bridges [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking endpoints [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking channels [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking events [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking recordings [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking playback [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking applications [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking bridges [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Got it! [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking bridgeId [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Got it! [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Finding handler for play [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking addChannel [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking removeChannel [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking mohStart [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking mohStop [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Checking play [Oct 10 14:43:08] DEBUG[16169] res_ari.c: Got it! [Oct 10 14:43:08] DEBUG[16169] bridge_roles.c: Set role 'announcer' [Oct 10 14:43:08] DEBUG[16169] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000007;1' [Oct 10 14:43:08] DEBUG[16169] res_stasis_playback.c: 1381434188.18: Sending play(sound:demo-congrats) command [Oct 10 14:43:08] DEBUG[16170] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910d154(Announcer/ARI-00000007;2) is joining [Oct 10 14:43:08] DEBUG[16171][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:43:08] DEBUG[16170] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x910d154(Announcer/ARI-00000007;2) [Oct 10 14:43:08] DEBUG[16170] bridge_roles.c: Set role 'announcer' [Oct 10 14:43:08] VERBOSE[16170] bridge_channel.c: -- Channel Announcer/ARI-00000007;2 joined 'softmix' base-bridge [Oct 10 14:43:08] DEBUG[16171][C-00000001] channel.c: Set channel Announcer/ARI-00000007;1 to write format gsm [Oct 10 14:43:08] DEBUG[16170] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:43:08] DEBUG[16170] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:43:08] DEBUG[16170] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:43:08] DEBUG[16170] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:43:08] DEBUG[16170] bridge.c: Chose bridge technology softmix [Oct 10 14:43:08] DEBUG[16170] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:43:08] DEBUG[16170] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000007;2 already has read format slin [Oct 10 14:43:08] DEBUG[16170] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000007;2 already has write format slin [Oct 10 14:43:08] DEBUG[16170] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910d154(Announcer/ARI-00000007;2) is joining softmix technology [Oct 10 14:43:08] DEBUG[16170] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:43:08] DEBUG[16170] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:43:08] DEBUG[16171][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:43:08] VERBOSE[16171][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:43:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:43:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:43:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:43:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:43:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:43:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:43:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:43:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:43:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for b5yzjVVJFYw2nD7oTDL7HPywKxJvorMJ - SUBSCRIBE (No RTP) [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:43:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:43:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog b5yzjVVJFYw2nD7oTDL7HPywKxJvorMJ [Oct 10 14:43:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:18] DEBUG[16164][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:43:18] DEBUG[16164][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:43:18] DEBUG[16164][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:43:18] DEBUG[16164][C-00000001] channel.c: Set channel Announcer/ARI-00000006;1 to write format slin [Oct 10 14:43:18] DEBUG[16164][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000006;1' [Oct 10 14:43:18] DEBUG[16163] bridge_channel.c: Setting 0xb2a892ac(Announcer/ARI-00000006;2) state from:0 to:1 [Oct 10 14:43:18] DEBUG[16163] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a892ac(Announcer/ARI-00000006;2) [Oct 10 14:43:18] VERBOSE[16163] bridge_channel.c: -- Channel Announcer/ARI-00000006;2 left 'softmix' base-bridge [Oct 10 14:43:18] DEBUG[16163] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a892ac(Announcer/ARI-00000006;2) is leaving softmix technology [Oct 10 14:43:18] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:43:18] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:43:18] DEBUG[16163] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:43:18] DEBUG[16163] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:43:18] DEBUG[16163] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:43:18] DEBUG[16163] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:43:18] DEBUG[16163] bridge.c: Chose bridge technology softmix [Oct 10 14:43:18] DEBUG[16163] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:43:18] DEBUG[16163] channel.c: Hanging up channel 'Announcer/ARI-00000006;2' [Oct 10 14:43:18] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:43:18] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:43:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:43:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:43:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:31] DEBUG[16174] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:43:31] DEBUG[16174] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:43:31] DEBUG[16174] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:43:31] DEBUG[16174] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:43:31] DEBUG[16174] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:43:31] DEBUG[16174] http.c: Match made with [ari] [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Finding handler for bridges [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking endpoints [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking channels [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking events [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking recordings [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking playback [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking applications [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking bridges [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Got it! [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking bridgeId [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Got it! [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Finding handler for play [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking addChannel [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking removeChannel [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking mohStart [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking mohStop [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Checking play [Oct 10 14:43:31] DEBUG[16174] res_ari.c: Got it! [Oct 10 14:43:31] DEBUG[16174] bridge_roles.c: Set role 'announcer' [Oct 10 14:43:31] DEBUG[16174] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000008;1' [Oct 10 14:43:31] DEBUG[16174] res_stasis_playback.c: 1381434211.20: Sending play(sound:demo-congrats) command [Oct 10 14:43:31] DEBUG[16177] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910bfac(Announcer/ARI-00000008;2) is joining [Oct 10 14:43:31] DEBUG[16178][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:43:31] DEBUG[16177] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x910bfac(Announcer/ARI-00000008;2) [Oct 10 14:43:31] DEBUG[16177] bridge_roles.c: Set role 'announcer' [Oct 10 14:43:31] VERBOSE[16177] bridge_channel.c: -- Channel Announcer/ARI-00000008;2 joined 'softmix' base-bridge [Oct 10 14:43:31] DEBUG[16177] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:43:31] DEBUG[16177] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:43:31] DEBUG[16177] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:43:31] DEBUG[16177] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:43:31] DEBUG[16177] bridge.c: Chose bridge technology softmix [Oct 10 14:43:31] DEBUG[16177] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:43:31] DEBUG[16177] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000008;2 already has read format slin [Oct 10 14:43:31] DEBUG[16177] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000008;2 already has write format slin [Oct 10 14:43:31] DEBUG[16177] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910bfac(Announcer/ARI-00000008;2) is joining softmix technology [Oct 10 14:43:31] DEBUG[16177] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:43:31] DEBUG[16177] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:43:31] DEBUG[16178][C-00000001] channel.c: Set channel Announcer/ARI-00000008;1 to write format gsm [Oct 10 14:43:31] DEBUG[16178][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:43:31] VERBOSE[16178][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:43:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:36] DEBUG[16171][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:43:36] DEBUG[16171][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:43:36] DEBUG[16171][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:43:36] DEBUG[16171][C-00000001] channel.c: Set channel Announcer/ARI-00000007;1 to write format slin [Oct 10 14:43:36] DEBUG[16171][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000007;1' [Oct 10 14:43:36] DEBUG[16170] bridge_channel.c: Setting 0x910d154(Announcer/ARI-00000007;2) state from:0 to:1 [Oct 10 14:43:36] DEBUG[16170] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0x910d154(Announcer/ARI-00000007;2) [Oct 10 14:43:36] VERBOSE[16170] bridge_channel.c: -- Channel Announcer/ARI-00000007;2 left 'softmix' base-bridge [Oct 10 14:43:36] DEBUG[16170] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910d154(Announcer/ARI-00000007;2) is leaving softmix technology [Oct 10 14:43:36] DEBUG[16170] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:43:36] DEBUG[16170] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:43:36] DEBUG[16170] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:43:36] DEBUG[16170] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:43:36] DEBUG[16170] bridge.c: Chose bridge technology softmix [Oct 10 14:43:36] DEBUG[16170] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:43:36] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:43:36] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:43:36] DEBUG[16170] channel.c: Hanging up channel 'Announcer/ARI-00000007;2' [Oct 10 14:43:36] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:43:36] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:43:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:39] DEBUG[16181] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:43:39] DEBUG[16181] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:43:39] DEBUG[16181] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:43:39] DEBUG[16181] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:43:39] DEBUG[16181] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:43:39] DEBUG[16181] http.c: Match made with [ari] [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Finding handler for bridges [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking endpoints [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking channels [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking events [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking recordings [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking playback [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking applications [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking bridges [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Got it! [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking bridgeId [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Got it! [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Finding handler for play [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking addChannel [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking removeChannel [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking mohStart [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking mohStop [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Checking play [Oct 10 14:43:39] DEBUG[16181] res_ari.c: Got it! [Oct 10 14:43:39] DEBUG[16181] bridge_roles.c: Set role 'announcer' [Oct 10 14:43:39] DEBUG[16181] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000009;1' [Oct 10 14:43:39] DEBUG[16181] res_stasis_playback.c: 1381434219.22: Sending play(sound:demo-congrats) command [Oct 10 14:43:39] DEBUG[16182] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2145194(Announcer/ARI-00000009;2) is joining [Oct 10 14:43:39] DEBUG[16182] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2145194(Announcer/ARI-00000009;2) [Oct 10 14:43:39] DEBUG[16182] bridge_roles.c: Set role 'announcer' [Oct 10 14:43:39] VERBOSE[16182] bridge_channel.c: -- Channel Announcer/ARI-00000009;2 joined 'softmix' base-bridge [Oct 10 14:43:39] DEBUG[16183][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:43:39] DEBUG[16182] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:43:39] DEBUG[16182] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:43:39] DEBUG[16182] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:43:39] DEBUG[16182] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:43:39] DEBUG[16182] bridge.c: Chose bridge technology softmix [Oct 10 14:43:39] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:43:39] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000009;2 already has read format slin [Oct 10 14:43:39] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000009;2 already has write format slin [Oct 10 14:43:39] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2145194(Announcer/ARI-00000009;2) is joining softmix technology [Oct 10 14:43:39] DEBUG[16182] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:43:39] DEBUG[16182] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:43:39] DEBUG[16183][C-00000001] channel.c: Set channel Announcer/ARI-00000009;1 to write format gsm [Oct 10 14:43:39] DEBUG[16183][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:43:39] VERBOSE[16183][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:43:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:43:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:43:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:43:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:43:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for zr9VHNogq85xJftcr-Mrh6OsPHqA9Uh9 - SUBSCRIBE (No RTP) [Oct 10 14:43:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:43:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:43:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:43:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:43:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog zr9VHNogq85xJftcr-Mrh6OsPHqA9Uh9 [Oct 10 14:43:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:43:59] DEBUG[16178][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:43:59] DEBUG[16178][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:43:59] DEBUG[16178][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:43:59] DEBUG[16178][C-00000001] channel.c: Set channel Announcer/ARI-00000008;1 to write format slin [Oct 10 14:43:59] DEBUG[16178][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000008;1' [Oct 10 14:43:59] DEBUG[16177] bridge_channel.c: Setting 0x910bfac(Announcer/ARI-00000008;2) state from:0 to:1 [Oct 10 14:43:59] DEBUG[16177] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0x910bfac(Announcer/ARI-00000008;2) [Oct 10 14:43:59] VERBOSE[16177] bridge_channel.c: -- Channel Announcer/ARI-00000008;2 left 'softmix' base-bridge [Oct 10 14:43:59] DEBUG[16177] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910bfac(Announcer/ARI-00000008;2) is leaving softmix technology [Oct 10 14:43:59] DEBUG[16177] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:43:59] DEBUG[16177] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:43:59] DEBUG[16177] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:43:59] DEBUG[16177] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:43:59] DEBUG[16177] bridge.c: Chose bridge technology softmix [Oct 10 14:43:59] DEBUG[16177] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:43:59] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:43:59] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:43:59] DEBUG[16177] channel.c: Hanging up channel 'Announcer/ARI-00000008;2' [Oct 10 14:43:59] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:43:59] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:44:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:07] DEBUG[16183][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:44:07] DEBUG[16183][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:44:07] DEBUG[16183][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:44:07] DEBUG[16183][C-00000001] channel.c: Set channel Announcer/ARI-00000009;1 to write format slin [Oct 10 14:44:07] DEBUG[16183][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000009;1' [Oct 10 14:44:07] DEBUG[16182] bridge_channel.c: Setting 0xb2145194(Announcer/ARI-00000009;2) state from:0 to:1 [Oct 10 14:44:07] DEBUG[16182] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2145194(Announcer/ARI-00000009;2) [Oct 10 14:44:07] VERBOSE[16182] bridge_channel.c: -- Channel Announcer/ARI-00000009;2 left 'softmix' base-bridge [Oct 10 14:44:07] DEBUG[16182] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2145194(Announcer/ARI-00000009;2) is leaving softmix technology [Oct 10 14:44:07] DEBUG[16182] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:44:07] DEBUG[16182] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:44:07] DEBUG[16182] bridge.c: Chose bridge technology native_rtp [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:44:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 876373701 to 836715949 due to a source change [Oct 10 14:44:07] DEBUG[16182] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:44:07] DEBUG[16182] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:44:07] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:44:07] DEBUG[16182] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:44:07] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:44:07] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:44:07] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:44:07] DEBUG[16182] channel.c: Hanging up channel 'Announcer/ARI-00000009;2' [Oct 10 14:44:07] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:44:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 2043565869 to 1064771639 due to a source change [Oct 10 14:44:07] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:44:07] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:44:07] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:44:07] DEBUG[16158][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:44:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:28] DEBUG[16205] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:44:28] DEBUG[16205] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:44:28] DEBUG[16205] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:44:28] DEBUG[16205] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:44:28] DEBUG[16205] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:44:28] DEBUG[16205] http.c: Match made with [ari] [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Finding handler for bridges [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking endpoints [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking channels [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking events [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking recordings [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking playback [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking applications [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking bridges [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Got it! [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking bridgeId [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Got it! [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Finding handler for play [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking addChannel [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking removeChannel [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking mohStart [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking mohStop [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Checking play [Oct 10 14:44:28] DEBUG[16205] res_ari.c: Got it! [Oct 10 14:44:28] DEBUG[16205] bridge_roles.c: Set role 'announcer' [Oct 10 14:44:28] DEBUG[16205] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000000a;1' [Oct 10 14:44:28] DEBUG[16205] res_stasis_playback.c: 1381434268.24: Sending play(sound:demo-congrats) command [Oct 10 14:44:28] DEBUG[16206] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910b744(Announcer/ARI-0000000a;2) is joining [Oct 10 14:44:28] DEBUG[16206] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x910b744(Announcer/ARI-0000000a;2) [Oct 10 14:44:28] DEBUG[16206] bridge_roles.c: Set role 'announcer' [Oct 10 14:44:28] VERBOSE[16206] bridge_channel.c: -- Channel Announcer/ARI-0000000a;2 joined 'native_rtp' base-bridge [Oct 10 14:44:28] DEBUG[16207][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:44:28] DEBUG[16206] bridge.c: Chose bridge technology softmix [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:44:28] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 836715949 to 1688076963 due to a source change [Oct 10 14:44:28] DEBUG[16206] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:44:28] DEBUG[16206] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:44:28] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1064771639 to 1828353781 due to a source change [Oct 10 14:44:28] DEBUG[16206] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:44:28] DEBUG[16206] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000a;2 already has read format slin [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000a;2 already has write format slin [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910b744(Announcer/ARI-0000000a;2) is joining softmix technology [Oct 10 14:44:28] DEBUG[16206] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:44:28] DEBUG[16206] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:44:28] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:44:28] DEBUG[16207][C-00000001] channel.c: Set channel Announcer/ARI-0000000a;1 to write format gsm [Oct 10 14:44:28] DEBUG[16207][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:44:28] VERBOSE[16207][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:44:28] DEBUG[16208][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:44:28] DEBUG[16208][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:44:28] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 170552, ms is 21339 [Oct 10 14:44:28] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 170552, ms is 21339 [Oct 10 14:44:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:38] DEBUG[16211] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:44:38] DEBUG[16211] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:44:38] DEBUG[16211] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:44:38] DEBUG[16211] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:44:38] DEBUG[16211] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:44:38] DEBUG[16211] http.c: Match made with [ari] [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Finding handler for bridges [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking endpoints [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking channels [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking events [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking recordings [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking playback [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking applications [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking bridges [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Got it! [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking bridgeId [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Got it! [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Finding handler for play [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking addChannel [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking removeChannel [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking mohStart [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking mohStop [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Checking play [Oct 10 14:44:38] DEBUG[16211] res_ari.c: Got it! [Oct 10 14:44:38] DEBUG[16211] bridge_roles.c: Set role 'announcer' [Oct 10 14:44:38] DEBUG[16211] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000000b;1' [Oct 10 14:44:38] DEBUG[16211] res_stasis_playback.c: 1381434278.26: Sending play(sound:demo-congrats) command [Oct 10 14:44:38] DEBUG[16212] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9110764(Announcer/ARI-0000000b;2) is joining [Oct 10 14:44:38] DEBUG[16212] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x9110764(Announcer/ARI-0000000b;2) [Oct 10 14:44:38] DEBUG[16212] bridge_roles.c: Set role 'announcer' [Oct 10 14:44:38] VERBOSE[16212] bridge_channel.c: -- Channel Announcer/ARI-0000000b;2 joined 'softmix' base-bridge [Oct 10 14:44:38] DEBUG[16213][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:44:38] DEBUG[16212] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:44:38] DEBUG[16212] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:44:38] DEBUG[16212] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:44:38] DEBUG[16212] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:44:38] DEBUG[16212] bridge.c: Chose bridge technology softmix [Oct 10 14:44:38] DEBUG[16212] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:44:38] DEBUG[16212] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000b;2 already has read format slin [Oct 10 14:44:38] DEBUG[16212] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000b;2 already has write format slin [Oct 10 14:44:38] DEBUG[16212] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9110764(Announcer/ARI-0000000b;2) is joining softmix technology [Oct 10 14:44:38] DEBUG[16213][C-00000001] channel.c: Set channel Announcer/ARI-0000000b;1 to write format gsm [Oct 10 14:44:38] DEBUG[16212] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:44:38] DEBUG[16212] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:44:38] DEBUG[16213][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:44:38] VERBOSE[16213][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:44:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:46] DEBUG[16217] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:44:46] DEBUG[16217] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:44:46] DEBUG[16217] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:44:46] DEBUG[16217] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:44:46] DEBUG[16217] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:44:46] DEBUG[16217] http.c: Match made with [ari] [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Finding handler for bridges [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking endpoints [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking channels [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking events [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking recordings [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking playback [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking applications [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking bridges [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Got it! [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking bridgeId [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Got it! [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Finding handler for play [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking addChannel [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking removeChannel [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking mohStart [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking mohStop [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Checking play [Oct 10 14:44:46] DEBUG[16217] res_ari.c: Got it! [Oct 10 14:44:46] DEBUG[16217] bridge_roles.c: Set role 'announcer' [Oct 10 14:44:46] DEBUG[16217] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000000c;1' [Oct 10 14:44:46] DEBUG[16217] res_stasis_playback.c: 1381434286.28: Sending play(sound:demo-congrats) command [Oct 10 14:44:46] DEBUG[16218] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2bf952c(Announcer/ARI-0000000c;2) is joining [Oct 10 14:44:46] DEBUG[16219][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:44:46] DEBUG[16218] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2bf952c(Announcer/ARI-0000000c;2) [Oct 10 14:44:46] DEBUG[16218] bridge_roles.c: Set role 'announcer' [Oct 10 14:44:46] VERBOSE[16218] bridge_channel.c: -- Channel Announcer/ARI-0000000c;2 joined 'softmix' base-bridge [Oct 10 14:44:46] DEBUG[16218] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:44:46] DEBUG[16218] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:44:46] DEBUG[16219][C-00000001] channel.c: Set channel Announcer/ARI-0000000c;1 to write format gsm [Oct 10 14:44:46] DEBUG[16218] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:44:46] DEBUG[16218] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:44:46] DEBUG[16218] bridge.c: Chose bridge technology softmix [Oct 10 14:44:46] DEBUG[16218] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:44:46] DEBUG[16218] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000c;2 already has read format slin [Oct 10 14:44:46] DEBUG[16219][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:44:46] DEBUG[16218] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000c;2 already has write format slin [Oct 10 14:44:46] DEBUG[16218] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2bf952c(Announcer/ARI-0000000c;2) is joining softmix technology [Oct 10 14:44:46] VERBOSE[16219][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:44:46] DEBUG[16218] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:44:46] DEBUG[16218] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:44:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:44:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:44:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for P-8US-6lZSLiJkfuBH63bQQyHxip.05A - SUBSCRIBE (No RTP) [Oct 10 14:44:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:44:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:44:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:44:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:44:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog P-8US-6lZSLiJkfuBH63bQQyHxip.05A [Oct 10 14:44:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:53] DEBUG[16222] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:44:53] DEBUG[16222] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:44:53] DEBUG[16222] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:44:53] DEBUG[16222] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:44:53] DEBUG[16222] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:44:53] DEBUG[16222] http.c: Match made with [ari] [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Finding handler for bridges [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking endpoints [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking channels [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking events [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking recordings [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking playback [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking applications [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking bridges [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Got it! [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking bridgeId [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Got it! [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Finding handler for play [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking addChannel [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking removeChannel [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking mohStart [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking mohStop [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Checking play [Oct 10 14:44:53] DEBUG[16222] res_ari.c: Got it! [Oct 10 14:44:53] DEBUG[16222] bridge_roles.c: Set role 'announcer' [Oct 10 14:44:53] DEBUG[16222] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000000d;1' [Oct 10 14:44:53] DEBUG[16222] res_stasis_playback.c: 1381434293.30: Sending play(sound:demo-congrats) command [Oct 10 14:44:53] DEBUG[16223] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2bfc9fc(Announcer/ARI-0000000d;2) is joining [Oct 10 14:44:53] DEBUG[16223] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2bfc9fc(Announcer/ARI-0000000d;2) [Oct 10 14:44:53] DEBUG[16224][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:44:53] DEBUG[16223] bridge_roles.c: Set role 'announcer' [Oct 10 14:44:53] VERBOSE[16223] bridge_channel.c: -- Channel Announcer/ARI-0000000d;2 joined 'softmix' base-bridge [Oct 10 14:44:53] DEBUG[16223] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:44:53] DEBUG[16223] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:44:53] DEBUG[16223] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:44:53] DEBUG[16223] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:44:53] DEBUG[16223] bridge.c: Chose bridge technology softmix [Oct 10 14:44:53] DEBUG[16223] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:44:53] DEBUG[16223] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000d;2 already has read format slin [Oct 10 14:44:53] DEBUG[16223] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000d;2 already has write format slin [Oct 10 14:44:53] DEBUG[16223] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2bfc9fc(Announcer/ARI-0000000d;2) is joining softmix technology [Oct 10 14:44:53] DEBUG[16223] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:44:53] DEBUG[16223] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:44:53] DEBUG[16224][C-00000001] channel.c: Set channel Announcer/ARI-0000000d;1 to write format gsm [Oct 10 14:44:53] DEBUG[16224][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:44:53] VERBOSE[16224][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:44:56] DEBUG[16207][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:44:56] DEBUG[16207][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:44:56] DEBUG[16207][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:44:56] DEBUG[16207][C-00000001] channel.c: Set channel Announcer/ARI-0000000a;1 to write format slin [Oct 10 14:44:56] DEBUG[16207][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000000a;1' [Oct 10 14:44:56] DEBUG[16206] bridge_channel.c: Setting 0x910b744(Announcer/ARI-0000000a;2) state from:0 to:1 [Oct 10 14:44:56] DEBUG[16206] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0x910b744(Announcer/ARI-0000000a;2) [Oct 10 14:44:56] VERBOSE[16206] bridge_channel.c: -- Channel Announcer/ARI-0000000a;2 left 'softmix' base-bridge [Oct 10 14:44:56] DEBUG[16206] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910b744(Announcer/ARI-0000000a;2) is leaving softmix technology [Oct 10 14:44:56] DEBUG[16206] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:44:56] DEBUG[16206] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:44:56] DEBUG[16206] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:44:56] DEBUG[16206] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:44:56] DEBUG[16206] bridge.c: Chose bridge technology softmix [Oct 10 14:44:56] DEBUG[16206] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:44:56] DEBUG[16206] channel.c: Hanging up channel 'Announcer/ARI-0000000a;2' [Oct 10 14:44:56] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:44:56] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:44:56] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:44:56] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:44:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:44:59] DEBUG[16227] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:44:59] DEBUG[16227] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:44:59] DEBUG[16227] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:44:59] DEBUG[16227] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:44:59] DEBUG[16227] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:44:59] DEBUG[16227] http.c: Match made with [ari] [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Finding handler for bridges [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking endpoints [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking channels [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking events [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking recordings [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking playback [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking applications [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking bridges [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Got it! [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking bridgeId [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Got it! [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Finding handler for play [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking addChannel [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking removeChannel [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking mohStart [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking mohStop [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Checking play [Oct 10 14:44:59] DEBUG[16227] res_ari.c: Got it! [Oct 10 14:44:59] DEBUG[16227] bridge_roles.c: Set role 'announcer' [Oct 10 14:44:59] DEBUG[16227] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000000e;1' [Oct 10 14:44:59] DEBUG[16227] res_stasis_playback.c: 1381434299.32: Sending play(sound:demo-congrats) command [Oct 10 14:44:59] DEBUG[16228] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2175c1c(Announcer/ARI-0000000e;2) is joining [Oct 10 14:44:59] DEBUG[16229][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:44:59] DEBUG[16228] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2175c1c(Announcer/ARI-0000000e;2) [Oct 10 14:44:59] DEBUG[16228] bridge_roles.c: Set role 'announcer' [Oct 10 14:44:59] VERBOSE[16228] bridge_channel.c: -- Channel Announcer/ARI-0000000e;2 joined 'softmix' base-bridge [Oct 10 14:44:59] DEBUG[16228] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:44:59] DEBUG[16228] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:44:59] DEBUG[16228] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:44:59] DEBUG[16228] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:44:59] DEBUG[16228] bridge.c: Chose bridge technology softmix [Oct 10 14:44:59] DEBUG[16228] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:44:59] DEBUG[16228] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000e;2 already has read format slin [Oct 10 14:44:59] DEBUG[16228] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000e;2 already has write format slin [Oct 10 14:44:59] DEBUG[16228] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2175c1c(Announcer/ARI-0000000e;2) is joining softmix technology [Oct 10 14:44:59] DEBUG[16228] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:44:59] DEBUG[16228] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:44:59] DEBUG[16229][C-00000001] channel.c: Set channel Announcer/ARI-0000000e;1 to write format gsm [Oct 10 14:44:59] DEBUG[16229][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:44:59] VERBOSE[16229][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:45:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:05] DEBUG[16236] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:45:05] DEBUG[16236] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:45:05] DEBUG[16236] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:45:05] DEBUG[16236] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:45:05] DEBUG[16236] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:45:05] DEBUG[16236] http.c: Match made with [ari] [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Finding handler for bridges [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking endpoints [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking channels [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking events [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking recordings [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking playback [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking applications [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking bridges [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Got it! [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking bridgeId [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Got it! [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Finding handler for play [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking addChannel [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking removeChannel [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking mohStart [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking mohStop [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Checking play [Oct 10 14:45:05] DEBUG[16236] res_ari.c: Got it! [Oct 10 14:45:05] DEBUG[16236] bridge_roles.c: Set role 'announcer' [Oct 10 14:45:05] DEBUG[16236] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000000f;1' [Oct 10 14:45:05] DEBUG[16236] res_stasis_playback.c: 1381434305.34: Sending play(sound:demo-congrats) command [Oct 10 14:45:05] DEBUG[16237] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910baf4(Announcer/ARI-0000000f;2) is joining [Oct 10 14:45:05] DEBUG[16237] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x910baf4(Announcer/ARI-0000000f;2) [Oct 10 14:45:05] DEBUG[16237] bridge_roles.c: Set role 'announcer' [Oct 10 14:45:05] VERBOSE[16237] bridge_channel.c: -- Channel Announcer/ARI-0000000f;2 joined 'softmix' base-bridge [Oct 10 14:45:05] DEBUG[16238][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:45:05] DEBUG[16237] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:05] DEBUG[16237] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:05] DEBUG[16237] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:05] DEBUG[16237] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:05] DEBUG[16237] bridge.c: Chose bridge technology softmix [Oct 10 14:45:05] DEBUG[16237] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:05] DEBUG[16237] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000f;2 already has read format slin [Oct 10 14:45:05] DEBUG[16237] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000000f;2 already has write format slin [Oct 10 14:45:05] DEBUG[16237] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910baf4(Announcer/ARI-0000000f;2) is joining softmix technology [Oct 10 14:45:05] DEBUG[16237] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:45:05] DEBUG[16237] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:45:05] DEBUG[16238][C-00000001] channel.c: Set channel Announcer/ARI-0000000f;1 to write format gsm [Oct 10 14:45:05] DEBUG[16238][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:45:05] VERBOSE[16238][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:45:06] DEBUG[16213][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:06] DEBUG[16213][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:06] DEBUG[16213][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:06] DEBUG[16213][C-00000001] channel.c: Set channel Announcer/ARI-0000000b;1 to write format slin [Oct 10 14:45:06] DEBUG[16213][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000000b;1' [Oct 10 14:45:06] DEBUG[16212] bridge_channel.c: Setting 0x9110764(Announcer/ARI-0000000b;2) state from:0 to:1 [Oct 10 14:45:06] DEBUG[16212] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0x9110764(Announcer/ARI-0000000b;2) [Oct 10 14:45:06] VERBOSE[16212] bridge_channel.c: -- Channel Announcer/ARI-0000000b;2 left 'softmix' base-bridge [Oct 10 14:45:06] DEBUG[16212] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9110764(Announcer/ARI-0000000b;2) is leaving softmix technology [Oct 10 14:45:06] DEBUG[16212] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:06] DEBUG[16212] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:06] DEBUG[16212] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:06] DEBUG[16212] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:06] DEBUG[16212] bridge.c: Chose bridge technology softmix [Oct 10 14:45:06] DEBUG[16212] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:06] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:06] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:06] DEBUG[16212] channel.c: Hanging up channel 'Announcer/ARI-0000000b;2' [Oct 10 14:45:06] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:06] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:10] DEBUG[16242] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:45:10] DEBUG[16242] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:45:10] DEBUG[16242] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:45:10] DEBUG[16242] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:45:10] DEBUG[16242] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:45:10] DEBUG[16242] http.c: Match made with [ari] [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Finding handler for bridges [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking endpoints [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking channels [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking events [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking recordings [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking playback [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking applications [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking bridges [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Got it! [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking bridgeId [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Got it! [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Finding handler for play [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking addChannel [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking removeChannel [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking mohStart [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking mohStop [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Checking play [Oct 10 14:45:10] DEBUG[16242] res_ari.c: Got it! [Oct 10 14:45:10] DEBUG[16242] bridge_roles.c: Set role 'announcer' [Oct 10 14:45:10] DEBUG[16242] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000010;1' [Oct 10 14:45:10] DEBUG[16242] res_stasis_playback.c: 1381434310.36: Sending play(sound:demo-congrats) command [Oct 10 14:45:10] DEBUG[16243] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a42bac(Announcer/ARI-00000010;2) is joining [Oct 10 14:45:10] DEBUG[16243] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a42bac(Announcer/ARI-00000010;2) [Oct 10 14:45:10] DEBUG[16243] bridge_roles.c: Set role 'announcer' [Oct 10 14:45:10] VERBOSE[16243] bridge_channel.c: -- Channel Announcer/ARI-00000010;2 joined 'softmix' base-bridge [Oct 10 14:45:10] DEBUG[16244][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:45:10] DEBUG[16243] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:10] DEBUG[16244][C-00000001] channel.c: Set channel Announcer/ARI-00000010;1 to write format gsm [Oct 10 14:45:10] DEBUG[16243] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:10] DEBUG[16243] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:10] DEBUG[16243] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:10] DEBUG[16244][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:45:10] DEBUG[16243] bridge.c: Chose bridge technology softmix [Oct 10 14:45:10] VERBOSE[16244][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:45:10] DEBUG[16243] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:10] DEBUG[16243] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000010;2 already has read format slin [Oct 10 14:45:10] DEBUG[16243] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000010;2 already has write format slin [Oct 10 14:45:10] DEBUG[16243] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a42bac(Announcer/ARI-00000010;2) is joining softmix technology [Oct 10 14:45:10] DEBUG[16243] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:45:10] DEBUG[16243] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:45:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:14] DEBUG[16247] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:45:14] DEBUG[16247] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:45:14] DEBUG[16247] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:45:14] DEBUG[16247] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:45:14] DEBUG[16247] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:45:14] DEBUG[16247] http.c: Match made with [ari] [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Finding handler for bridges [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking endpoints [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking channels [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking events [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking recordings [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking playback [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking applications [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking bridges [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Got it! [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking bridgeId [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Got it! [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Finding handler for play [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking addChannel [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking removeChannel [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking mohStart [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking mohStop [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Checking play [Oct 10 14:45:14] DEBUG[16247] res_ari.c: Got it! [Oct 10 14:45:14] DEBUG[16247] bridge_roles.c: Set role 'announcer' [Oct 10 14:45:14] DEBUG[16247] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000011;1' [Oct 10 14:45:14] DEBUG[16247] res_stasis_playback.c: 1381434314.38: Sending play(sound:demo-congrats) command [Oct 10 14:45:14] DEBUG[16248] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-00000011;2) is joining [Oct 10 14:45:14] DEBUG[16248] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a888fc(Announcer/ARI-00000011;2) [Oct 10 14:45:14] DEBUG[16248] bridge_roles.c: Set role 'announcer' [Oct 10 14:45:14] VERBOSE[16248] bridge_channel.c: -- Channel Announcer/ARI-00000011;2 joined 'softmix' base-bridge [Oct 10 14:45:14] DEBUG[16249][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:45:14] DEBUG[16249][C-00000001] channel.c: Set channel Announcer/ARI-00000011;1 to write format gsm [Oct 10 14:45:14] DEBUG[16248] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:14] DEBUG[16248] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:14] DEBUG[16248] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:14] DEBUG[16248] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:14] DEBUG[16248] bridge.c: Chose bridge technology softmix [Oct 10 14:45:14] DEBUG[16248] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:14] DEBUG[16248] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000011;2 already has read format slin [Oct 10 14:45:14] DEBUG[16248] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000011;2 already has write format slin [Oct 10 14:45:14] DEBUG[16248] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-00000011;2) is joining softmix technology [Oct 10 14:45:14] DEBUG[16248] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:45:14] DEBUG[16248] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:45:14] DEBUG[16249][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:45:14] VERBOSE[16249][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:45:14] DEBUG[16219][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:14] DEBUG[16219][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:14] DEBUG[16219][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:14] DEBUG[16219][C-00000001] channel.c: Set channel Announcer/ARI-0000000c;1 to write format slin [Oct 10 14:45:14] DEBUG[16219][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000000c;1' [Oct 10 14:45:14] DEBUG[16218] bridge_channel.c: Setting 0xb2bf952c(Announcer/ARI-0000000c;2) state from:0 to:1 [Oct 10 14:45:14] DEBUG[16218] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2bf952c(Announcer/ARI-0000000c;2) [Oct 10 14:45:14] VERBOSE[16218] bridge_channel.c: -- Channel Announcer/ARI-0000000c;2 left 'softmix' base-bridge [Oct 10 14:45:14] DEBUG[16218] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2bf952c(Announcer/ARI-0000000c;2) is leaving softmix technology [Oct 10 14:45:14] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:14] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:14] DEBUG[16218] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:14] DEBUG[16218] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:14] DEBUG[16218] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:14] DEBUG[16218] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:14] DEBUG[16218] bridge.c: Chose bridge technology softmix [Oct 10 14:45:14] DEBUG[16218] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:14] DEBUG[16218] channel.c: Hanging up channel 'Announcer/ARI-0000000c;2' [Oct 10 14:45:14] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:14] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:18] DEBUG[16252] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:45:18] DEBUG[16252] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:45:18] DEBUG[16252] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:45:18] DEBUG[16252] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:45:18] DEBUG[16252] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:45:18] DEBUG[16252] http.c: Match made with [ari] [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Finding handler for bridges [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking endpoints [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking channels [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking events [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking recordings [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking playback [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking applications [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking bridges [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Got it! [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking bridgeId [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Got it! [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Finding handler for play [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking addChannel [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking removeChannel [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking mohStart [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking mohStop [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Checking play [Oct 10 14:45:18] DEBUG[16252] res_ari.c: Got it! [Oct 10 14:45:18] DEBUG[16252] bridge_roles.c: Set role 'announcer' [Oct 10 14:45:18] DEBUG[16252] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000012;1' [Oct 10 14:45:18] DEBUG[16252] res_stasis_playback.c: 1381434318.40: Sending play(sound:demo-congrats) command [Oct 10 14:45:18] DEBUG[16253] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a970d4(Announcer/ARI-00000012;2) is joining [Oct 10 14:45:18] DEBUG[16253] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a970d4(Announcer/ARI-00000012;2) [Oct 10 14:45:18] DEBUG[16253] bridge_roles.c: Set role 'announcer' [Oct 10 14:45:18] DEBUG[16254][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:45:18] VERBOSE[16253] bridge_channel.c: -- Channel Announcer/ARI-00000012;2 joined 'softmix' base-bridge [Oct 10 14:45:18] DEBUG[16254][C-00000001] channel.c: Set channel Announcer/ARI-00000012;1 to write format gsm [Oct 10 14:45:18] DEBUG[16253] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:18] DEBUG[16253] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:18] DEBUG[16253] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:18] DEBUG[16253] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:18] DEBUG[16253] bridge.c: Chose bridge technology softmix [Oct 10 14:45:18] DEBUG[16253] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:18] DEBUG[16253] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000012;2 already has read format slin [Oct 10 14:45:18] DEBUG[16253] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000012;2 already has write format slin [Oct 10 14:45:18] DEBUG[16253] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a970d4(Announcer/ARI-00000012;2) is joining softmix technology [Oct 10 14:45:18] DEBUG[16253] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:45:18] DEBUG[16253] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:45:18] DEBUG[16254][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:45:18] VERBOSE[16254][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:45:21] DEBUG[16224][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:21] DEBUG[16224][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:21] DEBUG[16224][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:21] DEBUG[16224][C-00000001] channel.c: Set channel Announcer/ARI-0000000d;1 to write format slin [Oct 10 14:45:21] DEBUG[16224][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000000d;1' [Oct 10 14:45:21] DEBUG[16223] bridge_channel.c: Setting 0xb2bfc9fc(Announcer/ARI-0000000d;2) state from:0 to:1 [Oct 10 14:45:21] DEBUG[16223] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2bfc9fc(Announcer/ARI-0000000d;2) [Oct 10 14:45:21] VERBOSE[16223] bridge_channel.c: -- Channel Announcer/ARI-0000000d;2 left 'softmix' base-bridge [Oct 10 14:45:21] DEBUG[16223] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2bfc9fc(Announcer/ARI-0000000d;2) is leaving softmix technology [Oct 10 14:45:21] DEBUG[16223] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:21] DEBUG[16223] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:21] DEBUG[16223] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:21] DEBUG[16223] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:21] DEBUG[16223] bridge.c: Chose bridge technology softmix [Oct 10 14:45:21] DEBUG[16223] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:21] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:21] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:21] DEBUG[16223] channel.c: Hanging up channel 'Announcer/ARI-0000000d;2' [Oct 10 14:45:21] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:21] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:45:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:45:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:45:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:45:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:45:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:45:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:45:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:45:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for NwxzGLNXNqD4hGsRiXFXmvmrMbcKuMWY - SUBSCRIBE (No RTP) [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:45:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:45:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog NwxzGLNXNqD4hGsRiXFXmvmrMbcKuMWY [Oct 10 14:45:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:27] DEBUG[16229][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:27] DEBUG[16229][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:27] DEBUG[16229][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:27] DEBUG[16229][C-00000001] channel.c: Set channel Announcer/ARI-0000000e;1 to write format slin [Oct 10 14:45:27] DEBUG[16229][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000000e;1' [Oct 10 14:45:27] DEBUG[16228] bridge_channel.c: Setting 0xb2175c1c(Announcer/ARI-0000000e;2) state from:0 to:1 [Oct 10 14:45:27] DEBUG[16228] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2175c1c(Announcer/ARI-0000000e;2) [Oct 10 14:45:27] VERBOSE[16228] bridge_channel.c: -- Channel Announcer/ARI-0000000e;2 left 'softmix' base-bridge [Oct 10 14:45:27] DEBUG[16228] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2175c1c(Announcer/ARI-0000000e;2) is leaving softmix technology [Oct 10 14:45:27] DEBUG[16228] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:27] DEBUG[16228] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:27] DEBUG[16228] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:27] DEBUG[16228] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:27] DEBUG[16228] bridge.c: Chose bridge technology softmix [Oct 10 14:45:27] DEBUG[16228] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:27] DEBUG[16228] channel.c: Hanging up channel 'Announcer/ARI-0000000e;2' [Oct 10 14:45:27] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:27] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:27] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:27] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:33] DEBUG[16238][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:33] DEBUG[16238][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:33] DEBUG[16238][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:33] DEBUG[16238][C-00000001] channel.c: Set channel Announcer/ARI-0000000f;1 to write format slin [Oct 10 14:45:33] DEBUG[16238][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000000f;1' [Oct 10 14:45:33] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:33] DEBUG[16237] bridge_channel.c: Setting 0x910baf4(Announcer/ARI-0000000f;2) state from:0 to:1 [Oct 10 14:45:33] DEBUG[16237] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0x910baf4(Announcer/ARI-0000000f;2) [Oct 10 14:45:33] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:33] VERBOSE[16237] bridge_channel.c: -- Channel Announcer/ARI-0000000f;2 left 'softmix' base-bridge [Oct 10 14:45:33] DEBUG[16237] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x910baf4(Announcer/ARI-0000000f;2) is leaving softmix technology [Oct 10 14:45:33] DEBUG[16237] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:33] DEBUG[16237] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:33] DEBUG[16237] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:33] DEBUG[16237] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:33] DEBUG[16237] bridge.c: Chose bridge technology softmix [Oct 10 14:45:33] DEBUG[16237] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:33] DEBUG[16237] channel.c: Hanging up channel 'Announcer/ARI-0000000f;2' [Oct 10 14:45:33] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:33] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:38] DEBUG[16244][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:38] DEBUG[16244][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:38] DEBUG[16244][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:38] DEBUG[16244][C-00000001] channel.c: Set channel Announcer/ARI-00000010;1 to write format slin [Oct 10 14:45:38] DEBUG[16244][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000010;1' [Oct 10 14:45:38] DEBUG[16243] bridge_channel.c: Setting 0xb2a42bac(Announcer/ARI-00000010;2) state from:0 to:1 [Oct 10 14:45:38] DEBUG[16243] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a42bac(Announcer/ARI-00000010;2) [Oct 10 14:45:38] VERBOSE[16243] bridge_channel.c: -- Channel Announcer/ARI-00000010;2 left 'softmix' base-bridge [Oct 10 14:45:38] DEBUG[16243] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a42bac(Announcer/ARI-00000010;2) is leaving softmix technology [Oct 10 14:45:38] DEBUG[16243] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:38] DEBUG[16243] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:38] DEBUG[16243] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:38] DEBUG[16243] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:38] DEBUG[16243] bridge.c: Chose bridge technology softmix [Oct 10 14:45:38] DEBUG[16243] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:38] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:38] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:38] DEBUG[16243] channel.c: Hanging up channel 'Announcer/ARI-00000010;2' [Oct 10 14:45:38] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:38] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:39] DEBUG[16257] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:45:39] DEBUG[16257] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:45:39] DEBUG[16257] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:45:39] DEBUG[16257] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:45:39] DEBUG[16257] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:45:39] DEBUG[16257] http.c: Match made with [ari] [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Finding handler for bridges [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking endpoints [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking channels [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking events [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking recordings [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking playback [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking applications [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking bridges [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Got it! [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking bridgeId [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Got it! [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Finding handler for play [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking addChannel [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking removeChannel [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking mohStart [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking mohStop [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Checking play [Oct 10 14:45:39] DEBUG[16257] res_ari.c: Got it! [Oct 10 14:45:39] DEBUG[16257] bridge_roles.c: Set role 'announcer' [Oct 10 14:45:39] DEBUG[16257] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000013;1' [Oct 10 14:45:39] DEBUG[16257] res_stasis_playback.c: 1381434339.42: Sending play(sound:demo-congrats) command [Oct 10 14:45:39] DEBUG[16258] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a88eac(Announcer/ARI-00000013;2) is joining [Oct 10 14:45:39] DEBUG[16258] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a88eac(Announcer/ARI-00000013;2) [Oct 10 14:45:39] DEBUG[16258] bridge_roles.c: Set role 'announcer' [Oct 10 14:45:39] VERBOSE[16258] bridge_channel.c: -- Channel Announcer/ARI-00000013;2 joined 'softmix' base-bridge [Oct 10 14:45:39] DEBUG[16259][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:45:39] DEBUG[16259][C-00000001] channel.c: Set channel Announcer/ARI-00000013;1 to write format gsm [Oct 10 14:45:39] DEBUG[16259][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:45:39] VERBOSE[16259][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:45:39] DEBUG[16258] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:39] DEBUG[16258] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:39] DEBUG[16258] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:39] DEBUG[16258] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:39] DEBUG[16258] bridge.c: Chose bridge technology softmix [Oct 10 14:45:39] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:39] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000013;2 already has read format slin [Oct 10 14:45:39] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000013;2 already has write format slin [Oct 10 14:45:39] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a88eac(Announcer/ARI-00000013;2) is joining softmix technology [Oct 10 14:45:39] DEBUG[16258] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:45:39] DEBUG[16258] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:45:42] DEBUG[16249][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:42] DEBUG[16249][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:42] DEBUG[16249][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:42] DEBUG[16249][C-00000001] channel.c: Set channel Announcer/ARI-00000011;1 to write format slin [Oct 10 14:45:42] DEBUG[16249][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000011;1' [Oct 10 14:45:42] DEBUG[16248] bridge_channel.c: Setting 0xb2a888fc(Announcer/ARI-00000011;2) state from:0 to:1 [Oct 10 14:45:42] DEBUG[16248] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a888fc(Announcer/ARI-00000011;2) [Oct 10 14:45:42] VERBOSE[16248] bridge_channel.c: -- Channel Announcer/ARI-00000011;2 left 'softmix' base-bridge [Oct 10 14:45:42] DEBUG[16248] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-00000011;2) is leaving softmix technology [Oct 10 14:45:42] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:42] DEBUG[16248] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:42] DEBUG[16248] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:42] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:42] DEBUG[16248] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:42] DEBUG[16248] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:42] DEBUG[16248] bridge.c: Chose bridge technology softmix [Oct 10 14:45:42] DEBUG[16248] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:42] DEBUG[16248] channel.c: Hanging up channel 'Announcer/ARI-00000011;2' [Oct 10 14:45:42] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:42] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:45] DEBUG[16254][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:45] DEBUG[16254][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:45] DEBUG[16254][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:45:45] DEBUG[16254][C-00000001] channel.c: Set channel Announcer/ARI-00000012;1 to write format slin [Oct 10 14:45:45] DEBUG[16254][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000012;1' [Oct 10 14:45:45] DEBUG[16253] bridge_channel.c: Setting 0xb2a970d4(Announcer/ARI-00000012;2) state from:0 to:1 [Oct 10 14:45:45] DEBUG[16253] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a970d4(Announcer/ARI-00000012;2) [Oct 10 14:45:45] VERBOSE[16253] bridge_channel.c: -- Channel Announcer/ARI-00000012;2 left 'softmix' base-bridge [Oct 10 14:45:45] DEBUG[16253] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a970d4(Announcer/ARI-00000012;2) is leaving softmix technology [Oct 10 14:45:45] DEBUG[16253] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:45:45] DEBUG[16253] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:45:45] DEBUG[16253] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:45:45] DEBUG[16253] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:45:45] DEBUG[16253] bridge.c: Chose bridge technology softmix [Oct 10 14:45:45] DEBUG[16253] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:45:45] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:45] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:45] DEBUG[16253] channel.c: Hanging up channel 'Announcer/ARI-00000012;2' [Oct 10 14:45:45] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:45:45] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:45:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:45:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:45:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:45:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:45:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:45:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:45:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:45:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:45:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for fGiU1Fv3f3-hV8xLd4GRd6V5VtCkUMjl - SUBSCRIBE (No RTP) [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:45:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:45:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog fGiU1Fv3f3-hV8xLd4GRd6V5VtCkUMjl [Oct 10 14:45:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:45:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:45:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:45:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:07] DEBUG[16259][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:46:07] DEBUG[16259][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:46:07] DEBUG[16259][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:46:07] DEBUG[16259][C-00000001] channel.c: Set channel Announcer/ARI-00000013;1 to write format slin [Oct 10 14:46:07] DEBUG[16259][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000013;1' [Oct 10 14:46:07] DEBUG[16258] bridge_channel.c: Setting 0xb2a88eac(Announcer/ARI-00000013;2) state from:0 to:1 [Oct 10 14:46:07] DEBUG[16258] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a88eac(Announcer/ARI-00000013;2) [Oct 10 14:46:07] VERBOSE[16258] bridge_channel.c: -- Channel Announcer/ARI-00000013;2 left 'softmix' base-bridge [Oct 10 14:46:07] DEBUG[16258] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a88eac(Announcer/ARI-00000013;2) is leaving softmix technology [Oct 10 14:46:07] DEBUG[16258] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:46:07] DEBUG[16258] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:46:07] DEBUG[16258] bridge.c: Chose bridge technology native_rtp [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:46:07] DEBUG[16258] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:46:07] DEBUG[16258] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:46:07] DEBUG[16258] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:46:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1828353781 to 535931746 due to a source change [Oct 10 14:46:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1688076963 to 1443160650 due to a source change [Oct 10 14:46:07] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:46:07] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:46:07] DEBUG[16258] channel.c: Hanging up channel 'Announcer/ARI-00000013;2' [Oct 10 14:46:07] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:46:07] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:46:07] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:46:07] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:46:07] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:46:07] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:46:07] DEBUG[16208][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:46:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:46:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:46:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:35] DEBUG[16265] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:46:35] DEBUG[16265] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:46:35] DEBUG[16265] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:46:35] DEBUG[16265] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:46:35] DEBUG[16265] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:46:35] DEBUG[16265] http.c: Match made with [ari] [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Finding handler for bridges [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking endpoints [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking channels [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking events [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking recordings [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking playback [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking applications [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking bridges [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Got it! [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking bridgeId [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Got it! [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Finding handler for play [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking addChannel [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking removeChannel [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking mohStart [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking mohStop [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Checking play [Oct 10 14:46:35] DEBUG[16265] res_ari.c: Got it! [Oct 10 14:46:35] DEBUG[16265] bridge_roles.c: Set role 'announcer' [Oct 10 14:46:35] DEBUG[16265] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000014;1' [Oct 10 14:46:35] DEBUG[16265] res_stasis_playback.c: 1381434395.44: Sending play(sound:demo-congrats) command [Oct 10 14:46:35] DEBUG[16266] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x90f4b2c(Announcer/ARI-00000014;2) is joining [Oct 10 14:46:35] DEBUG[16266] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x90f4b2c(Announcer/ARI-00000014;2) [Oct 10 14:46:35] DEBUG[16266] bridge_roles.c: Set role 'announcer' [Oct 10 14:46:35] DEBUG[16267][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:46:35] VERBOSE[16266] bridge_channel.c: -- Channel Announcer/ARI-00000014;2 joined 'native_rtp' base-bridge [Oct 10 14:46:35] DEBUG[16267][C-00000001] channel.c: Set channel Announcer/ARI-00000014;1 to write format gsm [Oct 10 14:46:35] DEBUG[16267][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:46:35] VERBOSE[16267][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:46:35] DEBUG[16266] bridge.c: Chose bridge technology softmix [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:46:35] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1443160650 to 1350234815 due to a source change [Oct 10 14:46:35] DEBUG[16266] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:46:35] DEBUG[16266] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:46:35] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 535931746 to 116269328 due to a source change [Oct 10 14:46:35] DEBUG[16266] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:46:35] DEBUG[16266] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000014;2 already has read format slin [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000014;2 already has write format slin [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x90f4b2c(Announcer/ARI-00000014;2) is joining softmix technology [Oct 10 14:46:35] DEBUG[16266] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:46:35] DEBUG[16266] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:46:35] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:46:35] DEBUG[16268][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:46:35] DEBUG[16268][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:46:35] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 223048, ms is 27901 [Oct 10 14:46:35] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 223048, ms is 27901 [Oct 10 14:46:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:46:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:03] DEBUG[16267][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:47:03] DEBUG[16267][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:47:03] DEBUG[16267][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:47:03] DEBUG[16267][C-00000001] channel.c: Set channel Announcer/ARI-00000014;1 to write format slin [Oct 10 14:47:03] DEBUG[16267][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000014;1' [Oct 10 14:47:03] DEBUG[16266] bridge_channel.c: Setting 0x90f4b2c(Announcer/ARI-00000014;2) state from:0 to:1 [Oct 10 14:47:03] DEBUG[16266] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0x90f4b2c(Announcer/ARI-00000014;2) [Oct 10 14:47:03] VERBOSE[16266] bridge_channel.c: -- Channel Announcer/ARI-00000014;2 left 'softmix' base-bridge [Oct 10 14:47:03] DEBUG[16266] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x90f4b2c(Announcer/ARI-00000014;2) is leaving softmix technology [Oct 10 14:47:03] DEBUG[16266] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:47:03] DEBUG[16266] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:47:03] DEBUG[16266] bridge.c: Chose bridge technology native_rtp [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:47:03] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1350234815 to 1315479923 due to a source change [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:47:03] DEBUG[16266] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:47:03] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 116269328 to 2058357691 due to a source change [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:47:03] DEBUG[16266] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:47:03] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:47:03] DEBUG[16266] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:47:03] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:47:03] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:47:03] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:47:03] DEBUG[16266] channel.c: Hanging up channel 'Announcer/ARI-00000014;2' [Oct 10 14:47:03] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:47:03] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:47:03] DEBUG[16268][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:47:03] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:47:03] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:47:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:40] DEBUG[16274] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:47:40] DEBUG[16274] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:47:40] DEBUG[16274] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:47:40] DEBUG[16274] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:47:40] DEBUG[16274] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:47:40] DEBUG[16274] http.c: Match made with [ari] [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Finding handler for bridges [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking endpoints [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking channels [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking events [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking recordings [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking playback [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking applications [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking bridges [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Got it! [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking bridgeId [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Got it! [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Finding handler for play [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking addChannel [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking removeChannel [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking mohStart [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking mohStop [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Checking play [Oct 10 14:47:40] DEBUG[16274] res_ari.c: Got it! [Oct 10 14:47:40] DEBUG[16274] bridge_roles.c: Set role 'announcer' [Oct 10 14:47:40] DEBUG[16274] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000015;1' [Oct 10 14:47:40] DEBUG[16274] res_stasis_playback.c: 1381434460.46: Sending play(sound:demo-congrats) command [Oct 10 14:47:40] DEBUG[16275] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb214019c(Announcer/ARI-00000015;2) is joining [Oct 10 14:47:40] DEBUG[16275] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb214019c(Announcer/ARI-00000015;2) [Oct 10 14:47:40] DEBUG[16275] bridge_roles.c: Set role 'announcer' [Oct 10 14:47:40] DEBUG[16276][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:47:40] VERBOSE[16275] bridge_channel.c: -- Channel Announcer/ARI-00000015;2 joined 'native_rtp' base-bridge [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:47:40] DEBUG[16275] bridge.c: Chose bridge technology softmix [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:47:40] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1315479923 to 1642619209 due to a source change [Oct 10 14:47:40] DEBUG[16275] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:47:40] DEBUG[16275] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:47:40] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 2058357691 to 1592816139 due to a source change [Oct 10 14:47:40] DEBUG[16275] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:47:40] DEBUG[16275] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000015;2 already has read format slin [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000015;2 already has write format slin [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb214019c(Announcer/ARI-00000015;2) is joining softmix technology [Oct 10 14:47:40] DEBUG[16276][C-00000001] channel.c: Set channel Announcer/ARI-00000015;1 to write format gsm [Oct 10 14:47:40] DEBUG[16275] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:47:40] DEBUG[16275] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:47:40] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:47:40] DEBUG[16277][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:47:40] DEBUG[16277][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:47:40] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 299536, ms is 37462 [Oct 10 14:47:40] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 299536, ms is 37462 [Oct 10 14:47:40] DEBUG[16276][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:47:40] VERBOSE[16276][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:47:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:47:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:47:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:47:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:47:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:47:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:47:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:47:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:47:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for mUiqbJLHIgsPHq9w8V.XhvB.ANIm5C8i - SUBSCRIBE (No RTP) [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:47:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:47:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog mUiqbJLHIgsPHq9w8V.XhvB.ANIm5C8i [Oct 10 14:47:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:47:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:08] DEBUG[16276][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:48:08] DEBUG[16276][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:48:08] DEBUG[16276][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:48:08] DEBUG[16276][C-00000001] channel.c: Set channel Announcer/ARI-00000015;1 to write format slin [Oct 10 14:48:08] DEBUG[16276][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000015;1' [Oct 10 14:48:08] DEBUG[16275] bridge_channel.c: Setting 0xb214019c(Announcer/ARI-00000015;2) state from:0 to:1 [Oct 10 14:48:08] DEBUG[16275] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb214019c(Announcer/ARI-00000015;2) [Oct 10 14:48:08] VERBOSE[16275] bridge_channel.c: -- Channel Announcer/ARI-00000015;2 left 'softmix' base-bridge [Oct 10 14:48:08] DEBUG[16275] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb214019c(Announcer/ARI-00000015;2) is leaving softmix technology [Oct 10 14:48:08] DEBUG[16275] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:48:08] DEBUG[16275] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:48:08] DEBUG[16275] bridge.c: Chose bridge technology native_rtp [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:48:08] DEBUG[16275] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:48:08] DEBUG[16275] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:48:08] DEBUG[16275] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:48:08] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1592816139 to 188255692 due to a source change [Oct 10 14:48:08] DEBUG[16275] channel.c: Hanging up channel 'Announcer/ARI-00000015;2' [Oct 10 14:48:08] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:48:08] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:48:08] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1642619209 to 1997173432 due to a source change [Oct 10 14:48:08] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:48:08] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:48:08] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:48:08] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:48:08] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:48:08] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:48:08] DEBUG[16277][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:48:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:48:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:48:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:48:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:48:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:48:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:48:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:48:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:48:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 5yGO6WRJ1iDrl3qecp0aQdeeqK5nZygv - SUBSCRIBE (No RTP) [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:48:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:48:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog 5yGO6WRJ1iDrl3qecp0aQdeeqK5nZygv [Oct 10 14:48:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:20] DEBUG[16283] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:48:20] DEBUG[16283] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:48:20] DEBUG[16283] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:48:20] DEBUG[16283] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:48:20] DEBUG[16283] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:48:20] DEBUG[16283] http.c: Match made with [ari] [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Finding handler for bridges [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking endpoints [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking channels [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking events [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking recordings [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking playback [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking applications [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking bridges [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Got it! [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking bridgeId [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Got it! [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Finding handler for play [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking addChannel [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking removeChannel [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking mohStart [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking mohStop [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Checking play [Oct 10 14:48:20] DEBUG[16283] res_ari.c: Got it! [Oct 10 14:48:20] DEBUG[16283] bridge_roles.c: Set role 'announcer' [Oct 10 14:48:20] DEBUG[16283] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000016;1' [Oct 10 14:48:20] DEBUG[16283] res_stasis_playback.c: 1381434500.48: Sending play(sound:demo-congrats) command [Oct 10 14:48:20] DEBUG[16284] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-00000016;2) is joining [Oct 10 14:48:20] DEBUG[16284] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a888fc(Announcer/ARI-00000016;2) [Oct 10 14:48:20] DEBUG[16284] bridge_roles.c: Set role 'announcer' [Oct 10 14:48:20] VERBOSE[16284] bridge_channel.c: -- Channel Announcer/ARI-00000016;2 joined 'native_rtp' base-bridge [Oct 10 14:48:20] DEBUG[16285][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:48:20] DEBUG[16285][C-00000001] channel.c: Set channel Announcer/ARI-00000016;1 to write format gsm [Oct 10 14:48:20] DEBUG[16285][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:48:20] VERBOSE[16285][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:48:20] DEBUG[16284] bridge.c: Chose bridge technology softmix [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:48:20] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1997173432 to 1253298007 due to a source change [Oct 10 14:48:20] DEBUG[16284] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:48:20] DEBUG[16284] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:48:20] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 188255692 to 1671561593 due to a source change [Oct 10 14:48:20] DEBUG[16284] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:48:20] DEBUG[16284] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000016;2 already has read format slin [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000016;2 already has write format slin [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-00000016;2) is joining softmix technology [Oct 10 14:48:20] DEBUG[16284] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:48:20] DEBUG[16284] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:48:20] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:48:20] DEBUG[16286][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:48:20] DEBUG[16286][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:48:20] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 95144, ms is 11913 [Oct 10 14:48:20] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 95144, ms is 11913 [Oct 10 14:48:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:48:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:48:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:33] DEBUG[16289] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:48:33] DEBUG[16289] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:48:33] DEBUG[16289] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:48:33] DEBUG[16289] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:48:33] DEBUG[16289] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:48:33] DEBUG[16289] http.c: Match made with [ari] [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Finding handler for bridges [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking endpoints [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking channels [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking events [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking recordings [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking playback [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking applications [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking bridges [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Got it! [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking bridgeId [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Got it! [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Finding handler for play [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking addChannel [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking removeChannel [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking mohStart [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking mohStop [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Checking play [Oct 10 14:48:33] DEBUG[16289] res_ari.c: Got it! [Oct 10 14:48:33] DEBUG[16289] bridge_roles.c: Set role 'announcer' [Oct 10 14:48:33] DEBUG[16289] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000017;1' [Oct 10 14:48:33] DEBUG[16289] res_stasis_playback.c: 1381434513.50: Sending play(sound:demo-congrats) command [Oct 10 14:48:33] DEBUG[16290] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a40ccc(Announcer/ARI-00000017;2) is joining [Oct 10 14:48:33] DEBUG[16290] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a40ccc(Announcer/ARI-00000017;2) [Oct 10 14:48:33] DEBUG[16290] bridge_roles.c: Set role 'announcer' [Oct 10 14:48:33] VERBOSE[16290] bridge_channel.c: -- Channel Announcer/ARI-00000017;2 joined 'softmix' base-bridge [Oct 10 14:48:33] DEBUG[16291][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:48:33] DEBUG[16290] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:48:33] DEBUG[16290] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:48:33] DEBUG[16290] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:48:33] DEBUG[16290] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:48:33] DEBUG[16290] bridge.c: Chose bridge technology softmix [Oct 10 14:48:33] DEBUG[16290] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:48:33] DEBUG[16290] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000017;2 already has read format slin [Oct 10 14:48:33] DEBUG[16290] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000017;2 already has write format slin [Oct 10 14:48:33] DEBUG[16290] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a40ccc(Announcer/ARI-00000017;2) is joining softmix technology [Oct 10 14:48:33] DEBUG[16290] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:48:33] DEBUG[16290] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:48:33] DEBUG[16291][C-00000001] channel.c: Set channel Announcer/ARI-00000017;1 to write format gsm [Oct 10 14:48:33] DEBUG[16291][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:48:33] VERBOSE[16291][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:48:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:42] DEBUG[16294] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:48:42] DEBUG[16294] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:48:42] DEBUG[16294] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:48:42] DEBUG[16294] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:48:42] DEBUG[16294] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:48:42] DEBUG[16294] http.c: Match made with [ari] [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Finding handler for bridges [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking endpoints [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking channels [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking events [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking recordings [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking playback [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking applications [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking bridges [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Got it! [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking bridgeId [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Got it! [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Finding handler for play [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking addChannel [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking removeChannel [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking mohStart [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking mohStop [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Checking play [Oct 10 14:48:42] DEBUG[16294] res_ari.c: Got it! [Oct 10 14:48:42] DEBUG[16294] bridge_roles.c: Set role 'announcer' [Oct 10 14:48:42] DEBUG[16294] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000018;1' [Oct 10 14:48:42] DEBUG[16294] res_stasis_playback.c: 1381434522.52: Sending play(sound:demo-congrats) command [Oct 10 14:48:42] DEBUG[16295] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9224b8c(Announcer/ARI-00000018;2) is joining [Oct 10 14:48:42] DEBUG[16295] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x9224b8c(Announcer/ARI-00000018;2) [Oct 10 14:48:42] DEBUG[16295] bridge_roles.c: Set role 'announcer' [Oct 10 14:48:42] VERBOSE[16295] bridge_channel.c: -- Channel Announcer/ARI-00000018;2 joined 'softmix' base-bridge [Oct 10 14:48:42] DEBUG[16295] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:48:42] DEBUG[16295] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:48:42] DEBUG[16296][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:48:42] DEBUG[16295] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:48:42] DEBUG[16295] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:48:42] DEBUG[16295] bridge.c: Chose bridge technology softmix [Oct 10 14:48:42] DEBUG[16295] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:48:42] DEBUG[16296][C-00000001] channel.c: Set channel Announcer/ARI-00000018;1 to write format gsm [Oct 10 14:48:42] DEBUG[16295] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000018;2 already has read format slin [Oct 10 14:48:42] DEBUG[16295] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000018;2 already has write format slin [Oct 10 14:48:42] DEBUG[16295] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9224b8c(Announcer/ARI-00000018;2) is joining softmix technology [Oct 10 14:48:42] DEBUG[16295] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:48:42] DEBUG[16295] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:48:42] DEBUG[16296][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:48:42] VERBOSE[16296][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:48:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:48] DEBUG[16285][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:48:48] DEBUG[16285][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:48:48] DEBUG[16285][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:48:48] DEBUG[16285][C-00000001] channel.c: Set channel Announcer/ARI-00000016;1 to write format slin [Oct 10 14:48:48] DEBUG[16285][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000016;1' [Oct 10 14:48:48] DEBUG[16284] bridge_channel.c: Setting 0xb2a888fc(Announcer/ARI-00000016;2) state from:0 to:1 [Oct 10 14:48:48] DEBUG[16284] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a888fc(Announcer/ARI-00000016;2) [Oct 10 14:48:48] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:48:48] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:48:48] VERBOSE[16284] bridge_channel.c: -- Channel Announcer/ARI-00000016;2 left 'softmix' base-bridge [Oct 10 14:48:48] DEBUG[16284] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-00000016;2) is leaving softmix technology [Oct 10 14:48:48] DEBUG[16284] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:48:48] DEBUG[16284] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:48:48] DEBUG[16284] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:48:48] DEBUG[16284] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:48:48] DEBUG[16284] bridge.c: Chose bridge technology softmix [Oct 10 14:48:48] DEBUG[16284] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:48:48] DEBUG[16284] channel.c: Hanging up channel 'Announcer/ARI-00000016;2' [Oct 10 14:48:48] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:48:48] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:48:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:48:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:48:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:48:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:48:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for ASTzEp5FPlfw4ARVo747W7qr955JR-Tn - SUBSCRIBE (No RTP) [Oct 10 14:48:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:48:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:48:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:48:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:48:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog ASTzEp5FPlfw4ARVo747W7qr955JR-Tn [Oct 10 14:48:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:48:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:01] DEBUG[16291][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:01] DEBUG[16291][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:01] DEBUG[16291][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:01] DEBUG[16291][C-00000001] channel.c: Set channel Announcer/ARI-00000017;1 to write format slin [Oct 10 14:49:01] DEBUG[16291][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000017;1' [Oct 10 14:49:01] DEBUG[16290] bridge_channel.c: Setting 0xb2a40ccc(Announcer/ARI-00000017;2) state from:0 to:1 [Oct 10 14:49:01] DEBUG[16290] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a40ccc(Announcer/ARI-00000017;2) [Oct 10 14:49:01] VERBOSE[16290] bridge_channel.c: -- Channel Announcer/ARI-00000017;2 left 'softmix' base-bridge [Oct 10 14:49:01] DEBUG[16290] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a40ccc(Announcer/ARI-00000017;2) is leaving softmix technology [Oct 10 14:49:01] DEBUG[16290] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:49:01] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:49:01] DEBUG[16290] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:49:01] DEBUG[16290] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:49:01] DEBUG[16290] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:49:01] DEBUG[16290] bridge.c: Chose bridge technology softmix [Oct 10 14:49:01] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:49:01] DEBUG[16290] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:49:01] DEBUG[16290] channel.c: Hanging up channel 'Announcer/ARI-00000017;2' [Oct 10 14:49:01] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:49:01] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:49:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:03] DEBUG[16301] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:49:03] DEBUG[16301] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:49:03] DEBUG[16301] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:49:03] DEBUG[16301] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:49:03] DEBUG[16301] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:49:03] DEBUG[16301] http.c: Match made with [ari] [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Finding handler for bridges [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking endpoints [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking channels [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking events [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking recordings [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking playback [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking applications [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking bridges [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Got it! [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking bridgeId [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Got it! [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Finding handler for play [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking addChannel [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking removeChannel [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking mohStart [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking mohStop [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Checking play [Oct 10 14:49:03] DEBUG[16301] res_ari.c: Got it! [Oct 10 14:49:03] DEBUG[16301] bridge_roles.c: Set role 'announcer' [Oct 10 14:49:03] DEBUG[16301] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000019;1' [Oct 10 14:49:03] DEBUG[16301] res_stasis_playback.c: 1381434543.54: Sending play(sound:demo-congrats) command [Oct 10 14:49:03] DEBUG[16302] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-00000019;2) is joining [Oct 10 14:49:03] DEBUG[16303][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:49:03] DEBUG[16302] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a888fc(Announcer/ARI-00000019;2) [Oct 10 14:49:03] DEBUG[16302] bridge_roles.c: Set role 'announcer' [Oct 10 14:49:03] VERBOSE[16302] bridge_channel.c: -- Channel Announcer/ARI-00000019;2 joined 'softmix' base-bridge [Oct 10 14:49:03] DEBUG[16302] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:49:03] DEBUG[16302] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:49:03] DEBUG[16302] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:49:03] DEBUG[16302] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:49:03] DEBUG[16302] bridge.c: Chose bridge technology softmix [Oct 10 14:49:03] DEBUG[16302] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:49:03] DEBUG[16302] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000019;2 already has read format slin [Oct 10 14:49:03] DEBUG[16302] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000019;2 already has write format slin [Oct 10 14:49:03] DEBUG[16302] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-00000019;2) is joining softmix technology [Oct 10 14:49:03] DEBUG[16302] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:49:03] DEBUG[16302] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:49:03] DEBUG[16303][C-00000001] channel.c: Set channel Announcer/ARI-00000019;1 to write format gsm [Oct 10 14:49:03] DEBUG[16303][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:49:03] VERBOSE[16303][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:49:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:07] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:49:07] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:49:07] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:49:07] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:49:07] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:49:07] DEBUG[15619] chan_sip.c: Acked pending invite 107 [Oct 10 14:49:07] DEBUG[15619] chan_sip.c: Stopping retransmission on 'PLockzot5wXYYBQ89eWd18qPrTFlxMPl' of Request 107: Match Found [Oct 10 14:49:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:09] DEBUG[16307] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:49:09] DEBUG[16307] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:49:09] DEBUG[16307] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:49:09] DEBUG[16307] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:49:09] DEBUG[16307] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:49:09] DEBUG[16307] http.c: Match made with [ari] [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Finding handler for bridges [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking endpoints [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking channels [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking events [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking recordings [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking playback [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking applications [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking bridges [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Got it! [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking bridgeId [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Got it! [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Finding handler for play [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking addChannel [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking removeChannel [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking mohStart [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking mohStop [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Checking play [Oct 10 14:49:09] DEBUG[16307] res_ari.c: Got it! [Oct 10 14:49:09] DEBUG[16307] bridge_roles.c: Set role 'announcer' [Oct 10 14:49:09] DEBUG[16307] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000001a;1' [Oct 10 14:49:09] DEBUG[16307] res_stasis_playback.c: 1381434549.56: Sending play(sound:demo-congrats) command [Oct 10 14:49:09] DEBUG[16308] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2148c8c(Announcer/ARI-0000001a;2) is joining [Oct 10 14:49:09] DEBUG[16308] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2148c8c(Announcer/ARI-0000001a;2) [Oct 10 14:49:09] DEBUG[16308] bridge_roles.c: Set role 'announcer' [Oct 10 14:49:09] VERBOSE[16308] bridge_channel.c: -- Channel Announcer/ARI-0000001a;2 joined 'softmix' base-bridge [Oct 10 14:49:09] DEBUG[16309][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:49:09] DEBUG[16308] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:49:09] DEBUG[16308] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:49:09] DEBUG[16308] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:49:09] DEBUG[16308] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:49:09] DEBUG[16308] bridge.c: Chose bridge technology softmix [Oct 10 14:49:09] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:49:09] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001a;2 already has read format slin [Oct 10 14:49:09] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001a;2 already has write format slin [Oct 10 14:49:09] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2148c8c(Announcer/ARI-0000001a;2) is joining softmix technology [Oct 10 14:49:09] DEBUG[16308] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:49:09] DEBUG[16308] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:49:09] DEBUG[16309][C-00000001] channel.c: Set channel Announcer/ARI-0000001a;1 to write format gsm [Oct 10 14:49:09] DEBUG[16309][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:49:09] VERBOSE[16309][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:49:10] DEBUG[16296][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:10] DEBUG[16296][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:10] DEBUG[16296][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:10] DEBUG[16296][C-00000001] channel.c: Set channel Announcer/ARI-00000018;1 to write format slin [Oct 10 14:49:10] DEBUG[16296][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000018;1' [Oct 10 14:49:10] DEBUG[16295] bridge_channel.c: Setting 0x9224b8c(Announcer/ARI-00000018;2) state from:0 to:1 [Oct 10 14:49:10] DEBUG[16295] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0x9224b8c(Announcer/ARI-00000018;2) [Oct 10 14:49:10] VERBOSE[16295] bridge_channel.c: -- Channel Announcer/ARI-00000018;2 left 'softmix' base-bridge [Oct 10 14:49:10] DEBUG[16295] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9224b8c(Announcer/ARI-00000018;2) is leaving softmix technology [Oct 10 14:49:10] DEBUG[16295] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:49:10] DEBUG[16295] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:49:10] DEBUG[16295] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:49:10] DEBUG[16295] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:49:10] DEBUG[16295] bridge.c: Chose bridge technology softmix [Oct 10 14:49:10] DEBUG[16295] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:49:10] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:49:10] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:49:10] DEBUG[16295] channel.c: Hanging up channel 'Announcer/ARI-00000018;2' [Oct 10 14:49:10] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:49:10] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:49:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:29] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:49:29] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:49:29] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:49:29] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:49:29] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:49:29] DEBUG[15619] chan_sip.c: Acked pending invite 107 [Oct 10 14:49:29] DEBUG[15619] chan_sip.c: Stopping retransmission on 'EVj1DqKxYCA3vR-L.ceVXGwe5iq0.Ixi' of Request 107: Match Found [Oct 10 14:49:31] DEBUG[16303][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:31] DEBUG[16303][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:31] DEBUG[16303][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:31] DEBUG[16303][C-00000001] channel.c: Set channel Announcer/ARI-00000019;1 to write format slin [Oct 10 14:49:31] DEBUG[16303][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000019;1' [Oct 10 14:49:31] DEBUG[16302] bridge_channel.c: Setting 0xb2a888fc(Announcer/ARI-00000019;2) state from:0 to:1 [Oct 10 14:49:31] DEBUG[16302] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a888fc(Announcer/ARI-00000019;2) [Oct 10 14:49:31] VERBOSE[16302] bridge_channel.c: -- Channel Announcer/ARI-00000019;2 left 'softmix' base-bridge [Oct 10 14:49:31] DEBUG[16302] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-00000019;2) is leaving softmix technology [Oct 10 14:49:31] DEBUG[16302] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:49:31] DEBUG[16302] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:49:31] DEBUG[16302] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:49:31] DEBUG[16302] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:49:31] DEBUG[16302] bridge.c: Chose bridge technology softmix [Oct 10 14:49:31] DEBUG[16302] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:49:31] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:49:31] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:49:31] DEBUG[16302] channel.c: Hanging up channel 'Announcer/ARI-00000019;2' [Oct 10 14:49:31] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:49:31] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:49:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:33] DEBUG[15619] acl.c: For destination '10.24.17.17', our source address is '10.24.18.161'. [Oct 10 14:49:33] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:49:33] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 386d43afbe1b-td1bumlsie3r - REGISTER (No RTP) [Oct 10 14:49:33] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:49:33] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:49:33] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:49:33] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.17.17:5060 [Oct 10 14:49:33] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_C [Oct 10 14:49:33] DEBUG[15569] chan_sip.c: Checking device state for peer phone_C [Oct 10 14:49:33] DEBUG[15569] devicestate.c: Changing state for SIP/phone_C - state 1 (Not in use) [Oct 10 14:49:33] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_C' [Oct 10 14:49:33] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_C' [Oct 10 14:49:33] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_C' [Oct 10 14:49:33] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_C' has not changed from 'Not in use' [Oct 10 14:49:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:37] DEBUG[16309][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:37] DEBUG[16309][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:37] DEBUG[16309][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:49:37] DEBUG[16309][C-00000001] channel.c: Set channel Announcer/ARI-0000001a;1 to write format slin [Oct 10 14:49:37] DEBUG[16309][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000001a;1' [Oct 10 14:49:37] DEBUG[16308] bridge_channel.c: Setting 0xb2148c8c(Announcer/ARI-0000001a;2) state from:0 to:1 [Oct 10 14:49:37] DEBUG[16308] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2148c8c(Announcer/ARI-0000001a;2) [Oct 10 14:49:37] VERBOSE[16308] bridge_channel.c: -- Channel Announcer/ARI-0000001a;2 left 'softmix' base-bridge [Oct 10 14:49:37] DEBUG[16308] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2148c8c(Announcer/ARI-0000001a;2) is leaving softmix technology [Oct 10 14:49:37] DEBUG[16308] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:49:37] DEBUG[16308] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:49:37] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:49:37] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:49:37] DEBUG[16308] bridge.c: Chose bridge technology native_rtp [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:49:37] DEBUG[16308] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:49:37] DEBUG[16308] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:49:37] DEBUG[16308] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:49:37] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:49:37] DEBUG[16308] channel.c: Hanging up channel 'Announcer/ARI-0000001a;2' [Oct 10 14:49:37] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:49:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1671561593 to 286985491 due to a source change [Oct 10 14:49:37] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:49:37] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:49:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1253298007 to 1178311130 due to a source change [Oct 10 14:49:37] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:49:37] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:49:37] DEBUG[16286][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:49:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:49:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:49:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 7JkPJWA8dBRGKDrpPjOqFjwuR2KpCCYF - SUBSCRIBE (No RTP) [Oct 10 14:49:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:49:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:49:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:49:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:49:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog 7JkPJWA8dBRGKDrpPjOqFjwuR2KpCCYF [Oct 10 14:49:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:49:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:05] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog '386d43afbe1b-td1bumlsie3r' [Oct 10 14:50:05] DEBUG[15619] chan_sip.c: Destroying SIP dialog 386d43afbe1b-td1bumlsie3r [Oct 10 14:50:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:09] DEBUG[16315] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:50:09] DEBUG[16315] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:50:09] DEBUG[16315] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:50:09] DEBUG[16315] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:50:09] DEBUG[16315] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:50:09] DEBUG[16315] http.c: Match made with [ari] [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Finding handler for bridges [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking endpoints [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking channels [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking events [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking recordings [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking playback [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking applications [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking bridges [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Got it! [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking bridgeId [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Got it! [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Finding handler for play [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking addChannel [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking removeChannel [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking mohStart [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking mohStop [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Checking play [Oct 10 14:50:09] DEBUG[16315] res_ari.c: Got it! [Oct 10 14:50:09] DEBUG[16315] bridge_roles.c: Set role 'announcer' [Oct 10 14:50:09] DEBUG[16315] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000001b;1' [Oct 10 14:50:09] DEBUG[16315] res_stasis_playback.c: 1381434609.58: Sending play(sound:demo-congrats) command [Oct 10 14:50:09] DEBUG[16316] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-0000001b;2) is joining [Oct 10 14:50:09] DEBUG[16316] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a888fc(Announcer/ARI-0000001b;2) [Oct 10 14:50:09] DEBUG[16316] bridge_roles.c: Set role 'announcer' [Oct 10 14:50:09] VERBOSE[16316] bridge_channel.c: -- Channel Announcer/ARI-0000001b;2 joined 'native_rtp' base-bridge [Oct 10 14:50:09] DEBUG[16317][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:50:09] DEBUG[16317][C-00000001] channel.c: Set channel Announcer/ARI-0000001b;1 to write format gsm [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:50:09] DEBUG[16316] bridge.c: Chose bridge technology softmix [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:50:09] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1178311130 to 1021722968 due to a source change [Oct 10 14:50:09] DEBUG[16316] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:50:09] DEBUG[16316] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:50:09] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 286985491 to 1255712895 due to a source change [Oct 10 14:50:09] DEBUG[16317][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:50:09] DEBUG[16316] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:50:09] DEBUG[16316] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:50:09] VERBOSE[16317][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001b;2 already has read format slin [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001b;2 already has write format slin [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-0000001b;2) is joining softmix technology [Oct 10 14:50:09] DEBUG[16316] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:50:09] DEBUG[16316] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:50:09] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:50:09] DEBUG[16318][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:50:09] DEBUG[16318][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:50:09] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 254616, ms is 31847 [Oct 10 14:50:09] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 254616, ms is 31847 [Oct 10 14:50:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:16] DEBUG[16321] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:50:16] DEBUG[16321] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:50:16] DEBUG[16321] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:50:16] DEBUG[16321] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:50:16] DEBUG[16321] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:50:16] DEBUG[16321] http.c: Match made with [ari] [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Finding handler for bridges [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking endpoints [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking channels [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking events [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking recordings [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking playback [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking applications [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking bridges [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Got it! [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking bridgeId [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Got it! [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Finding handler for play [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking addChannel [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking removeChannel [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking mohStart [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking mohStop [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Checking play [Oct 10 14:50:16] DEBUG[16321] res_ari.c: Got it! [Oct 10 14:50:16] DEBUG[16321] bridge_roles.c: Set role 'announcer' [Oct 10 14:50:16] DEBUG[16321] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000001c;1' [Oct 10 14:50:16] DEBUG[16321] res_stasis_playback.c: 1381434616.60: Sending play(sound:demo-congrats) command [Oct 10 14:50:16] DEBUG[16322] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a429e4(Announcer/ARI-0000001c;2) is joining [Oct 10 14:50:16] DEBUG[16323][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:50:16] DEBUG[16322] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a429e4(Announcer/ARI-0000001c;2) [Oct 10 14:50:16] DEBUG[16322] bridge_roles.c: Set role 'announcer' [Oct 10 14:50:16] VERBOSE[16322] bridge_channel.c: -- Channel Announcer/ARI-0000001c;2 joined 'softmix' base-bridge [Oct 10 14:50:16] DEBUG[16322] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:50:16] DEBUG[16322] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:50:16] DEBUG[16322] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:50:16] DEBUG[16322] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:50:16] DEBUG[16322] bridge.c: Chose bridge technology softmix [Oct 10 14:50:16] DEBUG[16322] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:50:16] DEBUG[16322] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001c;2 already has read format slin [Oct 10 14:50:16] DEBUG[16322] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001c;2 already has write format slin [Oct 10 14:50:16] DEBUG[16322] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a429e4(Announcer/ARI-0000001c;2) is joining softmix technology [Oct 10 14:50:16] DEBUG[16322] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:50:16] DEBUG[16323][C-00000001] channel.c: Set channel Announcer/ARI-0000001c;1 to write format gsm [Oct 10 14:50:16] DEBUG[16322] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:50:16] DEBUG[16323][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:50:16] VERBOSE[16323][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:50:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:50:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:50:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:50:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:50:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:50:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:50:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:50:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:50:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for mXT1-92OtsxGzyz3acIZx0ml4DsenhBP - SUBSCRIBE (No RTP) [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:50:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:50:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog mXT1-92OtsxGzyz3acIZx0ml4DsenhBP [Oct 10 14:50:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:37] DEBUG[16317][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:50:37] DEBUG[16317][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:50:37] DEBUG[16317][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:50:37] DEBUG[16317][C-00000001] channel.c: Set channel Announcer/ARI-0000001b;1 to write format slin [Oct 10 14:50:37] DEBUG[16317][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000001b;1' [Oct 10 14:50:37] DEBUG[16316] bridge_channel.c: Setting 0xb2a888fc(Announcer/ARI-0000001b;2) state from:0 to:1 [Oct 10 14:50:37] DEBUG[16316] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a888fc(Announcer/ARI-0000001b;2) [Oct 10 14:50:37] VERBOSE[16316] bridge_channel.c: -- Channel Announcer/ARI-0000001b;2 left 'softmix' base-bridge [Oct 10 14:50:37] DEBUG[16316] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-0000001b;2) is leaving softmix technology [Oct 10 14:50:37] DEBUG[16316] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:50:37] DEBUG[16316] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:50:37] DEBUG[16316] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:50:37] DEBUG[16316] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:50:37] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:50:37] DEBUG[16316] bridge.c: Chose bridge technology softmix [Oct 10 14:50:37] DEBUG[16316] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:50:37] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:50:37] DEBUG[16316] channel.c: Hanging up channel 'Announcer/ARI-0000001b;2' [Oct 10 14:50:37] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:50:37] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:50:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:40] DEBUG[16326] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:50:40] DEBUG[16326] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:50:40] DEBUG[16326] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:50:40] DEBUG[16326] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:50:40] DEBUG[16326] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:50:40] DEBUG[16326] http.c: Match made with [ari] [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Finding handler for bridges [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking endpoints [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking channels [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking events [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking recordings [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking playback [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking applications [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking bridges [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Got it! [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking bridgeId [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Got it! [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Finding handler for play [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking addChannel [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking removeChannel [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking mohStart [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking mohStop [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Checking play [Oct 10 14:50:40] DEBUG[16326] res_ari.c: Got it! [Oct 10 14:50:40] DEBUG[16326] bridge_roles.c: Set role 'announcer' [Oct 10 14:50:40] DEBUG[16326] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000001d;1' [Oct 10 14:50:40] DEBUG[16326] res_stasis_playback.c: 1381434640.62: Sending play(sound:demo-congrats) command [Oct 10 14:50:40] DEBUG[16327] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb214eb3c(Announcer/ARI-0000001d;2) is joining [Oct 10 14:50:40] DEBUG[16327] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb214eb3c(Announcer/ARI-0000001d;2) [Oct 10 14:50:40] DEBUG[16327] bridge_roles.c: Set role 'announcer' [Oct 10 14:50:40] VERBOSE[16327] bridge_channel.c: -- Channel Announcer/ARI-0000001d;2 joined 'softmix' base-bridge [Oct 10 14:50:40] DEBUG[16328][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:50:40] DEBUG[16327] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:50:40] DEBUG[16327] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:50:40] DEBUG[16327] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:50:40] DEBUG[16327] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:50:40] DEBUG[16327] bridge.c: Chose bridge technology softmix [Oct 10 14:50:40] DEBUG[16327] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:50:40] DEBUG[16327] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001d;2 already has read format slin [Oct 10 14:50:40] DEBUG[16327] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001d;2 already has write format slin [Oct 10 14:50:40] DEBUG[16327] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb214eb3c(Announcer/ARI-0000001d;2) is joining softmix technology [Oct 10 14:50:40] DEBUG[16327] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:50:40] DEBUG[16327] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:50:40] DEBUG[16328][C-00000001] channel.c: Set channel Announcer/ARI-0000001d;1 to write format gsm [Oct 10 14:50:40] DEBUG[16328][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:50:40] VERBOSE[16328][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:50:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:44] DEBUG[15619] chan_sip.c: Session timer expired: 119 - iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX [Oct 10 14:50:44] DEBUG[15619] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Oct 10 14:50:44] DEBUG[15619] chan_sip.c: ** Our prefcodec: (nothing) [Oct 10 14:50:44] DEBUG[15619] chan_sip.c: -- Done with adding codecs to SDP [Oct 10 14:50:44] DEBUG[15619] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Oct 10 14:50:44] DEBUG[15619] chan_sip.c: Initializing already initialized SIP dialog iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX (presumably reinvite) [Oct 10 14:50:44] DEBUG[15619] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:50:44] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Oct 10 14:50:44] DEBUG[16323][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:50:44] DEBUG[16323][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:50:44] DEBUG[16323][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:50:44] DEBUG[16323][C-00000001] channel.c: Set channel Announcer/ARI-0000001c;1 to write format slin [Oct 10 14:50:44] DEBUG[16323][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000001c;1' [Oct 10 14:50:44] DEBUG[16322] bridge_channel.c: Setting 0xb2a429e4(Announcer/ARI-0000001c;2) state from:0 to:1 [Oct 10 14:50:44] DEBUG[16322] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a429e4(Announcer/ARI-0000001c;2) [Oct 10 14:50:44] VERBOSE[16322] bridge_channel.c: -- Channel Announcer/ARI-0000001c;2 left 'softmix' base-bridge [Oct 10 14:50:44] DEBUG[16322] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a429e4(Announcer/ARI-0000001c;2) is leaving softmix technology [Oct 10 14:50:44] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:50:44] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:50:44] DEBUG[16322] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:50:44] DEBUG[16322] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:50:44] DEBUG[16322] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:50:44] DEBUG[16322] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:50:44] DEBUG[16322] bridge.c: Chose bridge technology softmix [Oct 10 14:50:44] DEBUG[16322] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:50:44] DEBUG[16322] channel.c: Hanging up channel 'Announcer/ARI-0000001c;2' [Oct 10 14:50:44] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:50:44] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:50:44] DEBUG[15619][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Acked pending invite 104 [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Stopping retransmission on 'iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX' of Request 104: Match Found [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: SIP response 200 to RE-invite on outgoing call iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP o=- 76803233 76803236 IN IP4 10.24.19.97... OK. [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.19.97... OK. [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Oct 10 14:50:44] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 0 based on m type on 0xb2d91ca8 [Oct 10 14:50:44] DEBUG[15619][C-00000001] rtp_engine.c: Setting payload 96 based on m type on 0xb2d91ca8 [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4031 IN IP4 10.24.19.97... UNSUPPORTED OR FAILED. [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Oct 10 14:50:44] DEBUG[15619][C-00000001] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0xb2a0a374' [Oct 10 14:50:44] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 0 from 0xb2d91ca8 to 0xb2a0a49c [Oct 10 14:50:44] DEBUG[15619][C-00000001] rtp_engine.c: Copying payload 96 from 0xb2d91ca8 to 0xb2a0a49c [Oct 10 14:50:44] DEBUG[15619][C-00000001] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0xb2a0a374' [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: We're settling with these formats: (ulaw) [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: We have an owner, now see if we need to change this call [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Updating call counter for incoming call [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Session-Expires: 1800 [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Refresher: UAC [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Session timer stopped: 119 - iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Session timer started: 159 - iVV5IT3RuVdzNfGjfrCwxlKtCpnjZ.jX 900000ms [Oct 10 14:50:44] DEBUG[15619][C-00000001] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:50:44] DEBUG[15619][C-00000001] logger.c: CALL_ID [C-00000001] being removed from thread. [Oct 10 14:50:44] DEBUG[16086][C-00000001] res_rtp_asterisk.c: 0xb2cc9aa8 -- Probation learning mode pass with source address 10.24.19.97:4030 [Oct 10 14:50:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:50:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:50:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:50:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:50:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:50:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:50:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:50:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:50:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for WuGO.J3kXAOM5ZQQRywxi7qQWsa5QiTb - SUBSCRIBE (No RTP) [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:50:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:50:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog WuGO.J3kXAOM5ZQQRywxi7qQWsa5QiTb [Oct 10 14:50:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:51] DEBUG[16332] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:50:51] DEBUG[16332] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:50:51] DEBUG[16332] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:50:51] DEBUG[16332] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:50:51] DEBUG[16332] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:50:51] DEBUG[16332] http.c: Match made with [ari] [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Finding handler for bridges [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking endpoints [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking channels [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking events [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking recordings [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking playback [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking applications [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking bridges [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Got it! [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking bridgeId [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Got it! [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Finding handler for play [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking addChannel [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking removeChannel [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking mohStart [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking mohStop [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Checking play [Oct 10 14:50:51] DEBUG[16332] res_ari.c: Got it! [Oct 10 14:50:51] DEBUG[16332] bridge_roles.c: Set role 'announcer' [Oct 10 14:50:51] DEBUG[16332] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000001e;1' [Oct 10 14:50:51] DEBUG[16332] res_stasis_playback.c: 1381434651.64: Sending play(sound:demo-congrats) command [Oct 10 14:50:51] DEBUG[16333] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260708c(Announcer/ARI-0000001e;2) is joining [Oct 10 14:50:51] DEBUG[16333] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb260708c(Announcer/ARI-0000001e;2) [Oct 10 14:50:51] DEBUG[16334][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:50:51] DEBUG[16333] bridge_roles.c: Set role 'announcer' [Oct 10 14:50:51] VERBOSE[16333] bridge_channel.c: -- Channel Announcer/ARI-0000001e;2 joined 'softmix' base-bridge [Oct 10 14:50:51] DEBUG[16334][C-00000001] channel.c: Set channel Announcer/ARI-0000001e;1 to write format gsm [Oct 10 14:50:51] DEBUG[16334][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:50:51] VERBOSE[16334][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:50:51] DEBUG[16333] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:50:51] DEBUG[16333] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:50:51] DEBUG[16333] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:50:51] DEBUG[16333] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:50:51] DEBUG[16333] bridge.c: Chose bridge technology softmix [Oct 10 14:50:51] DEBUG[16333] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:50:51] DEBUG[16333] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001e;2 already has read format slin [Oct 10 14:50:51] DEBUG[16333] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001e;2 already has write format slin [Oct 10 14:50:51] DEBUG[16333] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260708c(Announcer/ARI-0000001e;2) is joining softmix technology [Oct 10 14:50:51] DEBUG[16333] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:50:51] DEBUG[16333] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:50:52] DEBUG[15619] chan_sip.c: Session timer expired: 123 - -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ [Oct 10 14:50:52] DEBUG[15619] chan_sip.c: ** Our capability: (ulaw) Video flag: True Text flag: True [Oct 10 14:50:52] DEBUG[15619] chan_sip.c: ** Our prefcodec: (nothing) [Oct 10 14:50:52] DEBUG[15619] chan_sip.c: -- Done with adding codecs to SDP [Oct 10 14:50:52] DEBUG[15619] chan_sip.c: Done building SDP. Settling with this capability: (ulaw) [Oct 10 14:50:52] DEBUG[15619] chan_sip.c: Initializing already initialized SIP dialog -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ (presumably reinvite) [Oct 10 14:50:52] DEBUG[15619] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:50:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 72 bytes [Oct 10 14:50:52] DEBUG[15619][C-00000002] logger.c: CALL_ID [C-00000002] bound to thread. [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Acked pending invite 104 [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Stopping retransmission on '-gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ' of Request 104: Match Found [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: SIP response 200 to RE-invite on outgoing call -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP o=- 76803241 76803244 IN IP4 10.24.18.165... OK. [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP s=digphn... UNSUPPORTED OR FAILED. [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 10.24.18.165... OK. [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing session-level SDP a=X-nat:0... UNSUPPORTED OR FAILED. [Oct 10 14:50:52] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0xb2d91ca8 [Oct 10 14:50:52] DEBUG[15619][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0xb2d91ca8 [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtcp:4013 IN IP4 10.24.18.165... UNSUPPORTED OR FAILED. [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK. [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 telephone-event/8000... OK. [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 0-15... UNSUPPORTED OR FAILED. [Oct 10 14:50:52] DEBUG[15619][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x912892c' [Oct 10 14:50:52] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 0 from 0xb2d91ca8 to 0x9128a54 [Oct 10 14:50:52] DEBUG[15619][C-00000002] rtp_engine.c: Copying payload 96 from 0xb2d91ca8 to 0x9128a54 [Oct 10 14:50:52] DEBUG[15619][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x912892c' [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: We're settling with these formats: (ulaw) [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: We have an owner, now see if we need to change this call [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Updating call counter for incoming call [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Session-Expires: 1800 [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Refresher: UAC [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Session timer stopped: 123 - -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Session timer started: 163 - -gTjFtKE.FM2fdd0FUWhFcXzbzJxgwwZ 900000ms [Oct 10 14:50:52] DEBUG[15619][C-00000002] chan_sip.c: Trying to put 'ACK sip:pho' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:50:52] DEBUG[15619][C-00000002] logger.c: CALL_ID [C-00000002] being removed from thread. [Oct 10 14:50:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: 0x912dec0 -- Probation learning mode pass with source address 10.24.18.165:4012 [Oct 10 14:50:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:53] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:50:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:50:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:50:58] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:03] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:08] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:08] DEBUG[16328][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:08] DEBUG[16328][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:08] DEBUG[16328][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:08] DEBUG[16328][C-00000001] channel.c: Set channel Announcer/ARI-0000001d;1 to write format slin [Oct 10 14:51:08] DEBUG[16328][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000001d;1' [Oct 10 14:51:08] DEBUG[16327] bridge_channel.c: Setting 0xb214eb3c(Announcer/ARI-0000001d;2) state from:0 to:1 [Oct 10 14:51:08] DEBUG[16327] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb214eb3c(Announcer/ARI-0000001d;2) [Oct 10 14:51:08] VERBOSE[16327] bridge_channel.c: -- Channel Announcer/ARI-0000001d;2 left 'softmix' base-bridge [Oct 10 14:51:08] DEBUG[16327] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb214eb3c(Announcer/ARI-0000001d;2) is leaving softmix technology [Oct 10 14:51:08] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:51:08] DEBUG[16327] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:51:08] DEBUG[16327] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:51:08] DEBUG[16327] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:51:08] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:51:08] DEBUG[16327] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:51:08] DEBUG[16327] bridge.c: Chose bridge technology softmix [Oct 10 14:51:08] DEBUG[16327] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:51:08] DEBUG[16327] channel.c: Hanging up channel 'Announcer/ARI-0000001d;2' [Oct 10 14:51:08] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:51:08] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:51:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:13] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:17] DEBUG[16339] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:51:17] DEBUG[16339] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:51:17] DEBUG[16339] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:51:17] DEBUG[16339] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:51:17] DEBUG[16339] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:51:17] DEBUG[16339] http.c: Match made with [ari] [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Finding handler for bridges [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking endpoints [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking channels [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking events [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking recordings [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking playback [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking applications [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking bridges [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Got it! [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking bridgeId [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Got it! [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Finding handler for play [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking addChannel [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking removeChannel [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking mohStart [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking mohStop [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Checking play [Oct 10 14:51:17] DEBUG[16339] res_ari.c: Got it! [Oct 10 14:51:17] DEBUG[16339] bridge_roles.c: Set role 'announcer' [Oct 10 14:51:17] DEBUG[16339] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-0000001f;1' [Oct 10 14:51:17] DEBUG[16339] res_stasis_playback.c: 1381434677.66: Sending play(sound:demo-congrats) command [Oct 10 14:51:17] DEBUG[16340] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-0000001f;2) is joining [Oct 10 14:51:17] DEBUG[16341][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:51:17] DEBUG[16340] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a888fc(Announcer/ARI-0000001f;2) [Oct 10 14:51:17] DEBUG[16340] bridge_roles.c: Set role 'announcer' [Oct 10 14:51:17] VERBOSE[16340] bridge_channel.c: -- Channel Announcer/ARI-0000001f;2 joined 'softmix' base-bridge [Oct 10 14:51:17] DEBUG[16341][C-00000001] channel.c: Set channel Announcer/ARI-0000001f;1 to write format gsm [Oct 10 14:51:17] DEBUG[16341][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:51:17] VERBOSE[16341][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:51:17] DEBUG[16340] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:51:17] DEBUG[16340] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:51:17] DEBUG[16340] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:51:17] DEBUG[16340] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:51:17] DEBUG[16340] bridge.c: Chose bridge technology softmix [Oct 10 14:51:17] DEBUG[16340] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:51:17] DEBUG[16340] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001f;2 already has read format slin [Oct 10 14:51:17] DEBUG[16340] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-0000001f;2 already has write format slin [Oct 10 14:51:17] DEBUG[16340] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-0000001f;2) is joining softmix technology [Oct 10 14:51:17] DEBUG[16340] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:51:17] DEBUG[16340] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:51:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:18] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:19] DEBUG[16334][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:19] DEBUG[16334][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:19] DEBUG[16334][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:19] DEBUG[16334][C-00000001] channel.c: Set channel Announcer/ARI-0000001e;1 to write format slin [Oct 10 14:51:19] DEBUG[16334][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000001e;1' [Oct 10 14:51:19] DEBUG[16333] bridge_channel.c: Setting 0xb260708c(Announcer/ARI-0000001e;2) state from:0 to:1 [Oct 10 14:51:19] DEBUG[16333] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb260708c(Announcer/ARI-0000001e;2) [Oct 10 14:51:19] VERBOSE[16333] bridge_channel.c: -- Channel Announcer/ARI-0000001e;2 left 'softmix' base-bridge [Oct 10 14:51:19] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:51:19] DEBUG[16333] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260708c(Announcer/ARI-0000001e;2) is leaving softmix technology [Oct 10 14:51:19] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:51:19] DEBUG[16333] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:51:19] DEBUG[16333] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:51:19] DEBUG[16333] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:51:19] DEBUG[16333] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:51:19] DEBUG[16333] bridge.c: Chose bridge technology softmix [Oct 10 14:51:19] DEBUG[16333] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:51:19] DEBUG[16333] channel.c: Hanging up channel 'Announcer/ARI-0000001e;2' [Oct 10 14:51:19] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:51:19] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:51:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:51:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:51:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:23] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:24] DEBUG[16344] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:51:24] DEBUG[16344] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:51:24] DEBUG[16344] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:51:24] DEBUG[16344] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:51:24] DEBUG[16344] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:51:24] DEBUG[16344] http.c: Match made with [ari] [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Finding handler for bridges [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking endpoints [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking channels [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking events [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking recordings [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking playback [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking applications [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking bridges [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Got it! [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking bridgeId [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Got it! [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Finding handler for play [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking addChannel [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking removeChannel [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking mohStart [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking mohStop [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Checking play [Oct 10 14:51:24] DEBUG[16344] res_ari.c: Got it! [Oct 10 14:51:24] DEBUG[16344] bridge_roles.c: Set role 'announcer' [Oct 10 14:51:24] DEBUG[16344] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000020;1' [Oct 10 14:51:24] DEBUG[16344] res_stasis_playback.c: 1381434684.68: Sending play(sound:demo-congrats) command [Oct 10 14:51:24] DEBUG[16345] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a96fac(Announcer/ARI-00000020;2) is joining [Oct 10 14:51:24] DEBUG[16345] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2a96fac(Announcer/ARI-00000020;2) [Oct 10 14:51:24] DEBUG[16346][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:51:24] DEBUG[16345] bridge_roles.c: Set role 'announcer' [Oct 10 14:51:24] VERBOSE[16345] bridge_channel.c: -- Channel Announcer/ARI-00000020;2 joined 'softmix' base-bridge [Oct 10 14:51:24] DEBUG[16346][C-00000001] channel.c: Set channel Announcer/ARI-00000020;1 to write format gsm [Oct 10 14:51:24] DEBUG[16346][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:51:24] VERBOSE[16346][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:51:24] DEBUG[16345] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:51:24] DEBUG[16345] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:51:24] DEBUG[16345] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:51:24] DEBUG[16345] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:51:24] DEBUG[16345] bridge.c: Chose bridge technology softmix [Oct 10 14:51:24] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:51:24] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000020;2 already has read format slin [Oct 10 14:51:24] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000020;2 already has write format slin [Oct 10 14:51:24] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a96fac(Announcer/ARI-00000020;2) is joining softmix technology [Oct 10 14:51:24] DEBUG[16345] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:51:24] DEBUG[16345] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:51:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:28] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:33] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:38] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:43] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:45] DEBUG[16341][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:45] DEBUG[16341][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:45] DEBUG[16341][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:45] DEBUG[16341][C-00000001] channel.c: Set channel Announcer/ARI-0000001f;1 to write format slin [Oct 10 14:51:45] DEBUG[16341][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-0000001f;1' [Oct 10 14:51:45] DEBUG[16340] bridge_channel.c: Setting 0xb2a888fc(Announcer/ARI-0000001f;2) state from:0 to:1 [Oct 10 14:51:45] DEBUG[16340] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a888fc(Announcer/ARI-0000001f;2) [Oct 10 14:51:45] VERBOSE[16340] bridge_channel.c: -- Channel Announcer/ARI-0000001f;2 left 'softmix' base-bridge [Oct 10 14:51:45] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:51:45] DEBUG[16340] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a888fc(Announcer/ARI-0000001f;2) is leaving softmix technology [Oct 10 14:51:45] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:51:45] DEBUG[16340] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:51:45] DEBUG[16340] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:51:45] DEBUG[16340] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:51:45] DEBUG[16340] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:51:45] DEBUG[16340] bridge.c: Chose bridge technology softmix [Oct 10 14:51:45] DEBUG[16340] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is already using the new technology. [Oct 10 14:51:45] DEBUG[16340] channel.c: Hanging up channel 'Announcer/ARI-0000001f;2' [Oct 10 14:51:45] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:51:45] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:51:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:48] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:52] DEBUG[16346][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:52] DEBUG[16346][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:52] DEBUG[16346][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:51:52] DEBUG[16346][C-00000001] channel.c: Set channel Announcer/ARI-00000020;1 to write format slin [Oct 10 14:51:52] DEBUG[16346][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000020;1' [Oct 10 14:51:52] DEBUG[16345] bridge_channel.c: Setting 0xb2a96fac(Announcer/ARI-00000020;2) state from:0 to:1 [Oct 10 14:51:52] DEBUG[16345] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2a96fac(Announcer/ARI-00000020;2) [Oct 10 14:51:52] VERBOSE[16345] bridge_channel.c: -- Channel Announcer/ARI-00000020;2 left 'softmix' base-bridge [Oct 10 14:51:52] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:51:52] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:51:52] DEBUG[16345] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2a96fac(Announcer/ARI-00000020;2) is leaving softmix technology [Oct 10 14:51:52] DEBUG[16345] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:51:52] DEBUG[16345] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:51:52] DEBUG[16345] bridge.c: Chose bridge technology native_rtp [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:51:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1021722968 to 477456445 due to a source change [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:51:52] DEBUG[16345] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:51:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1255712895 to 1225996429 due to a source change [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:51:52] DEBUG[16345] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:51:52] DEBUG[16345] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:51:52] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:51:52] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:51:52] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:51:52] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:51:52] DEBUG[16345] channel.c: Hanging up channel 'Announcer/ARI-00000020;2' [Oct 10 14:51:52] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:51:52] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:51:52] DEBUG[16318][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:51:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:53] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:57] DEBUG[16350] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:51:57] DEBUG[16350] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:51:57] DEBUG[16350] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:51:57] DEBUG[16350] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:51:57] DEBUG[16350] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:51:57] DEBUG[16350] http.c: Match made with [ari] [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Finding handler for bridges [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking endpoints [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking channels [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking events [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking recordings [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking playback [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking applications [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking bridges [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Got it! [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking bridgeId [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Got it! [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Finding handler for play [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking addChannel [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking removeChannel [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking mohStart [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking mohStop [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Checking play [Oct 10 14:51:57] DEBUG[16350] res_ari.c: Got it! [Oct 10 14:51:57] DEBUG[16350] bridge_roles.c: Set role 'announcer' [Oct 10 14:51:57] DEBUG[16350] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000021;1' [Oct 10 14:51:57] DEBUG[16350] res_stasis_playback.c: 1381434717.70: Sending play(sound:demo-congrats) command [Oct 10 14:51:57] DEBUG[16351] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260708c(Announcer/ARI-00000021;2) is joining [Oct 10 14:51:57] DEBUG[16352][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:51:57] DEBUG[16351] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb260708c(Announcer/ARI-00000021;2) [Oct 10 14:51:57] DEBUG[16351] bridge_roles.c: Set role 'announcer' [Oct 10 14:51:57] VERBOSE[16351] bridge_channel.c: -- Channel Announcer/ARI-00000021;2 joined 'native_rtp' base-bridge [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:51:57] DEBUG[16351] bridge.c: Chose bridge technology softmix [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:51:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 477456445 to 769513126 due to a source change [Oct 10 14:51:57] DEBUG[16351] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:51:57] DEBUG[16351] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:51:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1225996429 to 2107655384 due to a source change [Oct 10 14:51:57] DEBUG[16351] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:51:57] DEBUG[16351] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000021;2 already has read format slin [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000021;2 already has write format slin [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260708c(Announcer/ARI-00000021;2) is joining softmix technology [Oct 10 14:51:57] DEBUG[16351] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:51:57] DEBUG[16351] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:51:57] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:51:57] DEBUG[16352][C-00000001] channel.c: Set channel Announcer/ARI-00000021;1 to write format gsm [Oct 10 14:51:57] DEBUG[16352][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:51:57] VERBOSE[16352][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:51:57] DEBUG[16353][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:51:57] DEBUG[16353][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:51:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 42456, ms is 5327 [Oct 10 14:51:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 42456, ms is 5327 [Oct 10 14:51:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:51:58] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:03] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:25] DEBUG[16352][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:52:25] DEBUG[16352][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:52:25] DEBUG[16352][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:52:25] DEBUG[16352][C-00000001] channel.c: Set channel Announcer/ARI-00000021;1 to write format slin [Oct 10 14:52:25] DEBUG[16352][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000021;1' [Oct 10 14:52:25] DEBUG[16351] bridge_channel.c: Setting 0xb260708c(Announcer/ARI-00000021;2) state from:0 to:1 [Oct 10 14:52:25] DEBUG[16351] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb260708c(Announcer/ARI-00000021;2) [Oct 10 14:52:25] VERBOSE[16351] bridge_channel.c: -- Channel Announcer/ARI-00000021;2 left 'softmix' base-bridge [Oct 10 14:52:25] DEBUG[16351] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260708c(Announcer/ARI-00000021;2) is leaving softmix technology [Oct 10 14:52:25] DEBUG[16351] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:52:25] DEBUG[16351] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:52:25] DEBUG[16351] bridge.c: Chose bridge technology native_rtp [Oct 10 14:52:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:52:25] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:52:25] DEBUG[16351] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:52:25] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 769513126 to 1627461508 due to a source change [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:52:25] DEBUG[16351] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:52:25] DEBUG[16351] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:52:25] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 2107655384 to 1620820428 due to a source change [Oct 10 14:52:25] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:52:25] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:52:25] DEBUG[16351] channel.c: Hanging up channel 'Announcer/ARI-00000021;2' [Oct 10 14:52:25] DEBUG[16353][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:52:25] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:52:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:52:25] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:52:25] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:52:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:30] DEBUG[16358] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:52:30] DEBUG[16358] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:52:30] DEBUG[16358] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:52:30] DEBUG[16358] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:52:30] DEBUG[16358] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:52:30] DEBUG[16358] http.c: Match made with [ari] [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Finding handler for bridges [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking endpoints [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking channels [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking events [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking recordings [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking playback [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking applications [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking bridges [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Got it! [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking bridgeId [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Got it! [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Finding handler for play [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking addChannel [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking removeChannel [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking mohStart [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking mohStop [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Checking play [Oct 10 14:52:30] DEBUG[16358] res_ari.c: Got it! [Oct 10 14:52:30] DEBUG[16358] bridge_roles.c: Set role 'announcer' [Oct 10 14:52:30] DEBUG[16358] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000022;1' [Oct 10 14:52:30] DEBUG[16359] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x911ce3c(Announcer/ARI-00000022;2) is joining [Oct 10 14:52:30] DEBUG[16358] res_stasis_playback.c: 1381434750.72: Sending play(sound:demo-congrats) command [Oct 10 14:52:30] DEBUG[16359] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0x911ce3c(Announcer/ARI-00000022;2) [Oct 10 14:52:30] DEBUG[16359] bridge_roles.c: Set role 'announcer' [Oct 10 14:52:30] VERBOSE[16359] bridge_channel.c: -- Channel Announcer/ARI-00000022;2 joined 'native_rtp' base-bridge [Oct 10 14:52:30] DEBUG[16360][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:52:30] DEBUG[16359] bridge.c: Chose bridge technology softmix [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:52:30] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1627461508 to 1281147205 due to a source change [Oct 10 14:52:30] DEBUG[16359] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:52:30] DEBUG[16359] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:52:30] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1620820428 to 1410774027 due to a source change [Oct 10 14:52:30] DEBUG[16359] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:52:30] DEBUG[16359] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000022;2 already has read format slin [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000022;2 already has write format slin [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x911ce3c(Announcer/ARI-00000022;2) is joining softmix technology [Oct 10 14:52:30] DEBUG[16359] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:52:30] DEBUG[16359] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:52:30] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:52:30] DEBUG[16360][C-00000001] channel.c: Set channel Announcer/ARI-00000022;1 to write format gsm [Oct 10 14:52:30] DEBUG[16361][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:52:30] DEBUG[16361][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:52:30] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 40312, ms is 5059 [Oct 10 14:52:30] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 40312, ms is 5059 [Oct 10 14:52:30] DEBUG[16360][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:52:30] VERBOSE[16360][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:52:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:52:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:52:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:52:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:52:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:52:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:52:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:52:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:52:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for pSUowgd7dxkXIuusxm-f7fUHx3NAQuMN - SUBSCRIBE (No RTP) [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:52:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:52:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog pSUowgd7dxkXIuusxm-f7fUHx3NAQuMN [Oct 10 14:52:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:52:58] DEBUG[16360][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:52:58] DEBUG[16360][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:52:58] DEBUG[16360][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:52:58] DEBUG[16360][C-00000001] channel.c: Set channel Announcer/ARI-00000022;1 to write format slin [Oct 10 14:52:58] DEBUG[16360][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000022;1' [Oct 10 14:52:58] DEBUG[16359] bridge_channel.c: Setting 0x911ce3c(Announcer/ARI-00000022;2) state from:0 to:1 [Oct 10 14:52:58] DEBUG[16359] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0x911ce3c(Announcer/ARI-00000022;2) [Oct 10 14:52:58] VERBOSE[16359] bridge_channel.c: -- Channel Announcer/ARI-00000022;2 left 'softmix' base-bridge [Oct 10 14:52:58] DEBUG[16359] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x911ce3c(Announcer/ARI-00000022;2) is leaving softmix technology [Oct 10 14:52:58] DEBUG[16359] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:52:58] DEBUG[16359] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:52:58] DEBUG[16359] bridge.c: Chose bridge technology native_rtp [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:52:58] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1281147205 to 1479009947 due to a source change [Oct 10 14:52:58] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:52:58] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:52:58] DEBUG[16359] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:52:58] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1410774027 to 2046979909 due to a source change [Oct 10 14:52:58] DEBUG[16359] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:52:58] DEBUG[16359] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:52:58] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:52:58] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:52:58] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:52:58] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:52:58] DEBUG[16359] channel.c: Hanging up channel 'Announcer/ARI-00000022;2' [Oct 10 14:52:58] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:52:58] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:52:58] DEBUG[16361][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:53:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:53:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:53:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:53:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:53:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:53:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:53:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:53:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:53:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for coziCGN3KfOy3Tr.QYJWTjbKHet.-8bU - SUBSCRIBE (No RTP) [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:53:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:53:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog coziCGN3KfOy3Tr.QYJWTjbKHet.-8bU [Oct 10 14:53:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:53:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:53:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:53:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:53:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:53:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:53:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for n.wxR0Dv1KslxqD2le8KmzIl6dcjSPRm - SUBSCRIBE (No RTP) [Oct 10 14:53:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:53:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:53:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:53:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:53:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog n.wxR0Dv1KslxqD2le8KmzIl6dcjSPRm [Oct 10 14:53:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:53:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:42] DEBUG[16372] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:54:42] DEBUG[16372] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:54:42] DEBUG[16372] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:54:42] DEBUG[16372] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:54:42] DEBUG[16372] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:54:42] DEBUG[16372] http.c: Match made with [ari] [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Finding handler for bridges [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking endpoints [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking channels [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking events [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking recordings [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking playback [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking applications [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking bridges [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Got it! [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking bridgeId [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Got it! [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Finding handler for play [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking addChannel [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking removeChannel [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking mohStart [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking mohStop [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Checking play [Oct 10 14:54:42] DEBUG[16372] res_ari.c: Got it! [Oct 10 14:54:42] DEBUG[16372] bridge_roles.c: Set role 'announcer' [Oct 10 14:54:42] DEBUG[16372] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000023;1' [Oct 10 14:54:42] DEBUG[16372] res_stasis_playback.c: 1381434882.74: Sending play(sound:demo-congrats) command [Oct 10 14:54:42] DEBUG[16373] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb213fd24(Announcer/ARI-00000023;2) is joining [Oct 10 14:54:42] DEBUG[16373] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb213fd24(Announcer/ARI-00000023;2) [Oct 10 14:54:42] DEBUG[16374][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:54:42] DEBUG[16373] bridge_roles.c: Set role 'announcer' [Oct 10 14:54:42] VERBOSE[16373] bridge_channel.c: -- Channel Announcer/ARI-00000023;2 joined 'native_rtp' base-bridge [Oct 10 14:54:42] DEBUG[16374][C-00000001] channel.c: Set channel Announcer/ARI-00000023;1 to write format gsm [Oct 10 14:54:42] DEBUG[16374][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:54:42] VERBOSE[16374][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:54:42] DEBUG[16373] bridge.c: Chose bridge technology softmix [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:54:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1479009947 to 1429659116 due to a source change [Oct 10 14:54:42] DEBUG[16373] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:54:42] DEBUG[16373] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:54:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 2046979909 to 174283161 due to a source change [Oct 10 14:54:42] DEBUG[16373] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:54:42] DEBUG[16373] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000023;2 already has read format slin [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000023;2 already has write format slin [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb213fd24(Announcer/ARI-00000023;2) is joining softmix technology [Oct 10 14:54:42] DEBUG[16373] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:54:42] DEBUG[16373] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:54:42] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:54:42] DEBUG[16375][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:54:42] DEBUG[16375][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:54:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 831232, ms is 103924 [Oct 10 14:54:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 831232, ms is 103924 [Oct 10 14:54:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:54:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:54:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for ynQ7EQxaZHLKJp8xcum6wIVg6hYk58UB - SUBSCRIBE (No RTP) [Oct 10 14:54:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:54:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:54:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:54:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:54:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog ynQ7EQxaZHLKJp8xcum6wIVg6hYk58UB [Oct 10 14:54:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:54:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:10] DEBUG[16374][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:55:10] DEBUG[16374][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:55:10] DEBUG[16374][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:55:10] DEBUG[16374][C-00000001] channel.c: Set channel Announcer/ARI-00000023;1 to write format slin [Oct 10 14:55:10] DEBUG[16374][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000023;1' [Oct 10 14:55:10] DEBUG[16373] bridge_channel.c: Setting 0xb213fd24(Announcer/ARI-00000023;2) state from:0 to:1 [Oct 10 14:55:10] DEBUG[16373] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb213fd24(Announcer/ARI-00000023;2) [Oct 10 14:55:10] VERBOSE[16373] bridge_channel.c: -- Channel Announcer/ARI-00000023;2 left 'softmix' base-bridge [Oct 10 14:55:10] DEBUG[16373] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb213fd24(Announcer/ARI-00000023;2) is leaving softmix technology [Oct 10 14:55:10] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:55:10] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:55:10] DEBUG[16373] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:55:10] DEBUG[16373] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:55:10] DEBUG[16373] bridge.c: Chose bridge technology native_rtp [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:55:10] DEBUG[16373] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:55:10] DEBUG[16373] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:55:10] DEBUG[16373] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:55:10] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 174283161 to 1038243448 due to a source change [Oct 10 14:55:10] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1429659116 to 218284633 due to a source change [Oct 10 14:55:10] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:55:10] DEBUG[16373] channel.c: Hanging up channel 'Announcer/ARI-00000023;2' [Oct 10 14:55:10] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:55:10] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:55:10] DEBUG[16375][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:55:10] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:55:10] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:55:10] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:55:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:55:25] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:55:25] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:55:25] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:55:25] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:55:25] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:55:25] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:55:25] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:55:25] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for eLqGy41jlD7vy.0lpDKR1HgQY1vaDYet - SUBSCRIBE (No RTP) [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:55:25] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:55:25] DEBUG[15619] chan_sip.c: Destroying SIP dialog eLqGy41jlD7vy.0lpDKR1HgQY1vaDYet [Oct 10 14:55:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:55:47] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:55:47] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:55:47] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:55:47] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:55:47] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:55:47] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:55:47] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:55:47] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 44Z.wqsrZUXsJuFc9fKUjDK8x.hstXOs - SUBSCRIBE (No RTP) [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:55:47] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:55:47] DEBUG[15619] chan_sip.c: Destroying SIP dialog 44Z.wqsrZUXsJuFc9fKUjDK8x.hstXOs [Oct 10 14:55:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:55:57] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:55:57] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:55:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:19] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:56:19] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:56:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:56:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:52] DEBUG[16393] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:57:52] DEBUG[16393] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:57:52] DEBUG[16393] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:57:52] DEBUG[16393] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:57:52] DEBUG[16393] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:57:52] DEBUG[16393] http.c: Match made with [ari] [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Finding handler for bridges [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking endpoints [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking channels [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking events [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking recordings [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking playback [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking applications [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking bridges [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Got it! [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking bridgeId [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Got it! [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Finding handler for play [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking addChannel [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking removeChannel [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking mohStart [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking mohStop [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Checking play [Oct 10 14:57:52] DEBUG[16393] res_ari.c: Got it! [Oct 10 14:57:52] DEBUG[16393] bridge_roles.c: Set role 'announcer' [Oct 10 14:57:52] DEBUG[16393] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000024;1' [Oct 10 14:57:52] DEBUG[16393] res_stasis_playback.c: 1381435072.76: Sending play(sound:demo-congrats) command [Oct 10 14:57:52] DEBUG[16394] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2607f84(Announcer/ARI-00000024;2) is joining [Oct 10 14:57:52] DEBUG[16394] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb2607f84(Announcer/ARI-00000024;2) [Oct 10 14:57:52] DEBUG[16395][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:57:52] DEBUG[16394] bridge_roles.c: Set role 'announcer' [Oct 10 14:57:52] VERBOSE[16394] bridge_channel.c: -- Channel Announcer/ARI-00000024;2 joined 'native_rtp' base-bridge [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:57:52] DEBUG[16394] bridge.c: Chose bridge technology softmix [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:57:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 218284633 to 970988651 due to a source change [Oct 10 14:57:52] DEBUG[16394] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:57:52] DEBUG[16394] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:57:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1038243448 to 1824764372 due to a source change [Oct 10 14:57:52] DEBUG[16394] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:57:52] DEBUG[16394] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000024;2 already has read format slin [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000024;2 already has write format slin [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2607f84(Announcer/ARI-00000024;2) is joining softmix technology [Oct 10 14:57:52] DEBUG[16394] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:57:52] DEBUG[16394] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:57:52] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:57:52] DEBUG[16396][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:57:52] DEBUG[16396][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:57:52] DEBUG[16395][C-00000001] channel.c: Set channel Announcer/ARI-00000024;1 to write format gsm [Oct 10 14:57:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 1296544, ms is 162088 [Oct 10 14:57:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 1296544, ms is 162088 [Oct 10 14:57:52] DEBUG[16395][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:57:52] VERBOSE[16395][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:57:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:57:55] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 14:57:55] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 14:57:55] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 14:57:55] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 14:57:55] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 14:57:55] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 14:57:55] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 14:57:55] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for mTH3qvOmSPUYyt1vfJvCusEVBTuqEHKV - SUBSCRIBE (No RTP) [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:57:55] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:57:55] DEBUG[15619] chan_sip.c: Destroying SIP dialog mTH3qvOmSPUYyt1vfJvCusEVBTuqEHKV [Oct 10 14:57:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:57:58] DEBUG[16398] http.c: HTTP Request URI is /ari/playback/control?playbackid=931eeb4c-14dd-4957-b0ad-b3031c4a9a26&operation=restart&api_key=admin:secret [Oct 10 14:57:58] DEBUG[16398] http.c: match request [ari/playback/control] with handler [httpstatus] len 10 [Oct 10 14:57:58] DEBUG[16398] http.c: match request [ari/playback/control] with handler [phoneprov] len 9 [Oct 10 14:57:58] DEBUG[16398] http.c: match request [ari/playback/control] with handler [static] len 6 [Oct 10 14:57:58] DEBUG[16398] http.c: match request [ari/playback/control] with handler [ari] len 3 [Oct 10 14:57:58] DEBUG[16398] http.c: Match made with [ari] [Oct 10 14:57:58] DEBUG[16398] res_ari.c: Finding handler for playback [Oct 10 14:57:58] DEBUG[16398] res_ari.c: Checking endpoints [Oct 10 14:57:58] DEBUG[16398] res_ari.c: Checking channels [Oct 10 14:57:58] DEBUG[16398] res_ari.c: Checking events [Oct 10 14:57:58] DEBUG[16398] res_ari.c: Checking recordings [Oct 10 14:57:58] DEBUG[16398] res_ari.c: Checking playback [Oct 10 14:57:58] DEBUG[16398] res_ari.c: Got it! [Oct 10 14:57:58] DEBUG[16398] res_ari.c: Finding handler for control [Oct 10 14:57:58] DEBUG[16398] res_ari.c: Checking playbackId [Oct 10 14:57:58] DEBUG[16398] res_ari.c: Got it! [Oct 10 14:58:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:58:17] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 14:58:17] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 14:58:17] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 14:58:17] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 14:58:17] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 14:58:17] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 14:58:17] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 14:58:17] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for 2-imrUlZQHRJJYPHTAaypsCunOd1629O - SUBSCRIBE (No RTP) [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:58:17] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:58:17] DEBUG[15619] chan_sip.c: Destroying SIP dialog 2-imrUlZQHRJJYPHTAaypsCunOd1629O [Oct 10 14:58:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:20] DEBUG[16395][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:58:20] DEBUG[16395][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:58:20] DEBUG[16395][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:58:20] DEBUG[16395][C-00000001] channel.c: Set channel Announcer/ARI-00000024;1 to write format slin [Oct 10 14:58:20] DEBUG[16395][C-00000001] channel.c: Hanging up channel 'Announcer/ARI-00000024;1' [Oct 10 14:58:20] DEBUG[16394] bridge_channel.c: Setting 0xb2607f84(Announcer/ARI-00000024;2) state from:0 to:1 [Oct 10 14:58:20] DEBUG[16394] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pulling 0xb2607f84(Announcer/ARI-00000024;2) [Oct 10 14:58:20] VERBOSE[16394] bridge_channel.c: -- Channel Announcer/ARI-00000024;2 left 'softmix' base-bridge [Oct 10 14:58:20] DEBUG[16394] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb2607f84(Announcer/ARI-00000024;2) is leaving softmix technology [Oct 10 14:58:20] DEBUG[16394] dahdi/bridge_native_dahdi.c: Channel 'SIP/phone_A-00000001' is not DAHDI. [Oct 10 14:58:20] DEBUG[16394] dahdi/bridge_native_dahdi.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Cannot use native DAHDI. Channel 'SIP/phone_A-00000001' not compatible. [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge technology native_dahdi is not compatible with properties of existing bridge. [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge technology softmix does not have any capabilities we want. [Oct 10 14:58:20] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:58:20] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 2 (In use) [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge technology simple_bridge has less preference than native_rtp (50 <= 90). Skipping. [Oct 10 14:58:20] DEBUG[16394] bridge.c: Chose bridge technology native_rtp [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology constructor [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology stop [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving softmix technology (dummy) [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining native_rtp technology [Oct 10 14:58:20] DEBUG[16394] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving softmix technology (dummy) [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining native_rtp technology [Oct 10 14:58:20] DEBUG[16394] bridge_native_rtp.c: Locally RTP bridged 'SIP/phone_A-00000001' and 'SIP/phone_B-00000002' in stack [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology start [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: deferring softmix technology destructor [Oct 10 14:58:20] DEBUG[16394] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: queueing action type:13 sub:1000 [Oct 10 14:58:20] DEBUG[16394] channel.c: Hanging up channel 'Announcer/ARI-00000024;2' [Oct 10 14:58:20] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 970988651 to 1725761301 due to a source change [Oct 10 14:58:20] DEBUG[16086][C-00000001] channel.c: SIP/phone_A-00000001: Dropping redundant connected line update "Phone B" <1002>. [Oct 10 14:58:20] DEBUG[15567][C-00000001] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology destructor (deferred, dummy) [Oct 10 14:58:20] DEBUG[15567][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: Waiting for mixing thread to die. [Oct 10 14:58:20] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for Announcer - ARI [Oct 10 14:58:20] DEBUG[15569] devicestate.c: Changing state for Announcer/ARI - state 0 (Unknown) [Oct 10 14:58:20] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 1824764372 to 756103515 due to a source change [Oct 10 14:58:20] DEBUG[16099][C-00000002] channel.c: SIP/phone_B-00000002: Dropping redundant connected line update "Phone A" <1001>. [Oct 10 14:58:20] DEBUG[16396][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: stopping mixing thread [Oct 10 14:58:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:27] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'QODzSz-932QKcSoWl6XMcphUiebV2olm' [Oct 10 14:58:27] DEBUG[15619] chan_sip.c: Destroying SIP dialog QODzSz-932QKcSoWl6XMcphUiebV2olm [Oct 10 14:58:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:49] DEBUG[15619] chan_sip.c: Auto destroying SIP dialog 'Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI' [Oct 10 14:58:49] DEBUG[15619] chan_sip.c: Destroying SIP dialog Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI [Oct 10 14:58:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:54] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 14:58:54] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:58:54] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for O-Iej3ETpEDiTINNAHJqkdO3UH08Wb-O - SUBSCRIBE (No RTP) [Oct 10 14:58:54] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:58:54] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 14:58:54] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:58:54] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 14:58:54] DEBUG[15619] chan_sip.c: Destroying SIP dialog O-Iej3ETpEDiTINNAHJqkdO3UH08Wb-O [Oct 10 14:58:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:58:58] DEBUG[16406] http.c: HTTP Request URI is /ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play?media=sound:demo-congrats&api_key=admin:secret [Oct 10 14:58:58] DEBUG[16406] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [httpstatus] len 10 [Oct 10 14:58:58] DEBUG[16406] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [phoneprov] len 9 [Oct 10 14:58:58] DEBUG[16406] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [static] len 6 [Oct 10 14:58:58] DEBUG[16406] http.c: match request [ari/bridges/f7c9433b-711e-4fdf-990e-7ea806eed848/play] with handler [ari] len 3 [Oct 10 14:58:58] DEBUG[16406] http.c: Match made with [ari] [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Finding handler for bridges [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking endpoints [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking channels [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking events [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking recordings [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking playback [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking applications [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking bridges [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Got it! [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Finding handler for f7c9433b-711e-4fdf-990e-7ea806eed848 [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking bridgeId [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Got it! [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Finding handler for play [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking addChannel [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking removeChannel [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking mohStart [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking mohStop [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Checking play [Oct 10 14:58:58] DEBUG[16406] res_ari.c: Got it! [Oct 10 14:58:58] DEBUG[16406] bridge_roles.c: Set role 'announcer' [Oct 10 14:58:58] DEBUG[16406] ari/resource_bridges.c: Created announcer channel 'Announcer/ARI-00000025;1' [Oct 10 14:58:58] DEBUG[16406] res_stasis_playback.c: 1381435138.78: Sending play(sound:demo-congrats) command [Oct 10 14:58:58] DEBUG[16407] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260708c(Announcer/ARI-00000025;2) is joining [Oct 10 14:58:58] DEBUG[16407] bridge_channel.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: pushing 0xb260708c(Announcer/ARI-00000025;2) [Oct 10 14:58:58] DEBUG[16407] bridge_roles.c: Set role 'announcer' [Oct 10 14:58:58] DEBUG[16408][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:58:58] VERBOSE[16407] bridge_channel.c: -- Channel Announcer/ARI-00000025;2 joined 'native_rtp' base-bridge [Oct 10 14:58:58] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format gsm [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge technology native_dahdi does not have any capabilities we want. [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge technology native_rtp does not have any capabilities we want. [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge technology holding_bridge does not have any capabilities we want. [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge technology simple_bridge does not have any capabilities we want. [Oct 10 14:58:58] DEBUG[16407] bridge.c: Chose bridge technology softmix [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology constructor [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology stop [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is leaving native_rtp technology (dummy) [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has read format slin [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_A-00000001 already has write format slin [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9245d9c(SIP/phone_A-00000001) is joining softmix technology [Oct 10 14:58:58] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Changing ssrc from 1725761301 to 899393015 due to a source change [Oct 10 14:58:58] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:58:58] VERBOSE[16408][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:58:58] DEBUG[16407] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:58:58] DEBUG[16407] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is leaving native_rtp technology (dummy) [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has read format slin [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel SIP/phone_B-00000002 already has write format slin [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0x9248004(SIP/phone_B-00000002) is joining softmix technology [Oct 10 14:58:58] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Changing ssrc from 756103515 to 580216799 due to a source change [Oct 10 14:58:58] DEBUG[16407] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:58:58] DEBUG[16407] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000025;2 already has read format slin [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848 is happy that channel Announcer/ARI-00000025;2 already has write format slin [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: 0xb260708c(Announcer/ARI-00000025;2) is joining softmix technology [Oct 10 14:58:58] DEBUG[16407] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Oct 10 14:58:58] DEBUG[16407] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling softmix technology start [Oct 10 14:58:58] DEBUG[16407] bridge.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: calling native_rtp technology destructor [Oct 10 14:58:58] DEBUG[16409][C-00000001] logger.c: CALL_ID [C-00000001] bound to thread. [Oct 10 14:58:58] DEBUG[16409][C-00000001] bridge_softmix.c: Bridge f7c9433b-711e-4fdf-990e-7ea806eed848: starting mixing thread [Oct 10 14:58:58] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 304608, ms is 38096 [Oct 10 14:58:58] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 304608, ms is 38096 [Oct 10 14:59:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:02] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:59:02] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:59:02] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:59:02] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_A [Oct 10 14:59:02] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:59:02] DEBUG[15619] chan_sip.c: Acked pending invite 108 [Oct 10 14:59:02] DEBUG[15619] chan_sip.c: Stopping retransmission on 'PLockzot5wXYYBQ89eWd18qPrTFlxMPl' of Request 108: Match Found [Oct 10 14:59:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:02] DEBUG[16412] http.c: HTTP Request URI is /ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control?playbackid=83fd10fa-6381-4ade-b55b-9427ac6a8065&operation=restart&api_key=admin:secret [Oct 10 14:59:02] DEBUG[16412] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [httpstatus] len 10 [Oct 10 14:59:02] DEBUG[16412] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [phoneprov] len 9 [Oct 10 14:59:02] DEBUG[16412] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [static] len 6 [Oct 10 14:59:02] DEBUG[16412] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [ari] len 3 [Oct 10 14:59:02] DEBUG[16412] http.c: Match made with [ari] [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Finding handler for playback [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Checking endpoints [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Checking channels [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Checking events [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Checking recordings [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Checking playback [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Got it! [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Finding handler for 83fd10fa-6381-4ade-b55b-9427ac6a8065 [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Checking playbackId [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Got it! [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Finding handler for control [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Checking control [Oct 10 14:59:02] DEBUG[16412] res_ari.c: Got it! [Oct 10 14:59:02] DEBUG[16408][C-00000001] app.c: we'll restart the stream here at next loop [Oct 10 14:59:02] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:59:02] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:59:02] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format slin [Oct 10 14:59:02] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format gsm [Oct 10 14:59:02] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:59:02] VERBOSE[16408][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:59:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:10] DEBUG[16415] http.c: HTTP Request URI is /ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control?playbackid=83fd10fa-6381-4ade-b55b-9427ac6a8065&operation=restart&api_key=admin:secret [Oct 10 14:59:10] DEBUG[16415] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [httpstatus] len 10 [Oct 10 14:59:10] DEBUG[16415] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [phoneprov] len 9 [Oct 10 14:59:10] DEBUG[16415] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [static] len 6 [Oct 10 14:59:10] DEBUG[16415] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [ari] len 3 [Oct 10 14:59:10] DEBUG[16415] http.c: Match made with [ari] [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Finding handler for playback [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Checking endpoints [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Checking channels [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Checking events [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Checking recordings [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Checking playback [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Got it! [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Finding handler for 83fd10fa-6381-4ade-b55b-9427ac6a8065 [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Checking playbackId [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Got it! [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Finding handler for control [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Checking control [Oct 10 14:59:10] DEBUG[16415] res_ari.c: Got it! [Oct 10 14:59:10] DEBUG[16408][C-00000001] app.c: we'll restart the stream here at next loop [Oct 10 14:59:10] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:59:10] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:59:10] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format slin [Oct 10 14:59:10] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format gsm [Oct 10 14:59:10] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:59:10] VERBOSE[16408][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:59:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:12] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:17] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:17] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:22] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:22] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:24] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:59:24] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:59:24] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:59:24] DEBUG[15619] devicestate.c: No provider found, checking channel drivers for PJSIP - phone_B [Oct 10 14:59:24] DEBUG[15619] chan_sip.c: Trying to put 'NOTIFY sip:' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 14:59:24] DEBUG[15619] chan_sip.c: Acked pending invite 108 [Oct 10 14:59:24] DEBUG[15619] chan_sip.c: Stopping retransmission on 'EVj1DqKxYCA3vR-L.ceVXGwe5iq0.Ixi' of Request 108: Match Found [Oct 10 14:59:27] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:27] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:32] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:32] DEBUG[16417] http.c: HTTP Request URI is /ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control?playbackid=83fd10fa-6381-4ade-b55b-9427ac6a8065&operation=restart&api_key=admin:secret [Oct 10 14:59:32] DEBUG[16417] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [httpstatus] len 10 [Oct 10 14:59:32] DEBUG[16417] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [phoneprov] len 9 [Oct 10 14:59:32] DEBUG[16417] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [static] len 6 [Oct 10 14:59:32] DEBUG[16417] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [ari] len 3 [Oct 10 14:59:32] DEBUG[16417] http.c: Match made with [ari] [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Finding handler for playback [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Checking endpoints [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Checking channels [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Checking events [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Checking recordings [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Checking playback [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Got it! [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Finding handler for 83fd10fa-6381-4ade-b55b-9427ac6a8065 [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Checking playbackId [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Got it! [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Finding handler for control [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Checking control [Oct 10 14:59:32] DEBUG[16417] res_ari.c: Got it! [Oct 10 14:59:32] DEBUG[16408][C-00000001] app.c: we'll restart the stream here at next loop [Oct 10 14:59:32] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:59:32] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:59:32] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format slin [Oct 10 14:59:32] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format gsm [Oct 10 14:59:32] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:59:32] VERBOSE[16408][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:59:32] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:37] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:37] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:42] DEBUG[16419] http.c: HTTP Request URI is /ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control?playbackid=83fd10fa-6381-4ade-b55b-9427ac6a8065&operation=restartzzzzz&api_key=admin:secret [Oct 10 14:59:42] DEBUG[16419] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [httpstatus] len 10 [Oct 10 14:59:42] DEBUG[16419] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [phoneprov] len 9 [Oct 10 14:59:42] DEBUG[16419] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [static] len 6 [Oct 10 14:59:42] DEBUG[16419] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [ari] len 3 [Oct 10 14:59:42] DEBUG[16419] http.c: Match made with [ari] [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Finding handler for playback [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Checking endpoints [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Checking channels [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Checking events [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Checking recordings [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Checking playback [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Got it! [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Finding handler for 83fd10fa-6381-4ade-b55b-9427ac6a8065 [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Checking playbackId [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Got it! [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Finding handler for control [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Checking control [Oct 10 14:59:42] DEBUG[16419] res_ari.c: Got it! [Oct 10 14:59:42] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:42] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:47] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:47] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:48] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 14:59:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 14:59:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for JYjgynARO1weCqtA7F-hyIF7ATyqfGBC - SUBSCRIBE (No RTP) [Oct 10 14:59:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 14:59:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 14:59:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 14:59:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 14:59:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog JYjgynARO1weCqtA7F-hyIF7ATyqfGBC [Oct 10 14:59:52] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:52] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:55] DEBUG[16422] http.c: HTTP Request URI is /ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control?playbackid=83fd10fa-6381-4ade-b55b-9427ac6a8065&operation=restart&api_key=admin:secret [Oct 10 14:59:55] DEBUG[16422] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [httpstatus] len 10 [Oct 10 14:59:55] DEBUG[16422] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [phoneprov] len 9 [Oct 10 14:59:55] DEBUG[16422] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [static] len 6 [Oct 10 14:59:55] DEBUG[16422] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [ari] len 3 [Oct 10 14:59:55] DEBUG[16422] http.c: Match made with [ari] [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Finding handler for playback [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Checking endpoints [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Checking channels [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Checking events [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Checking recordings [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Checking playback [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Got it! [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Finding handler for 83fd10fa-6381-4ade-b55b-9427ac6a8065 [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Checking playbackId [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Got it! [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Finding handler for control [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Checking control [Oct 10 14:59:55] DEBUG[16422] res_ari.c: Got it! [Oct 10 14:59:55] DEBUG[16408][C-00000001] app.c: we'll restart the stream here at next loop [Oct 10 14:59:55] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:59:55] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 14:59:55] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format slin [Oct 10 14:59:55] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format gsm [Oct 10 14:59:55] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 14:59:55] VERBOSE[16408][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 14:59:57] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 14:59:57] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:02] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:02] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:07] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:07] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:11] DEBUG[16429] http.c: HTTP Request URI is /ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control?playbackid=83fd10fa-6381-4ade-b55b-9427ac6a8065&operation=&api_key=admin:secret [Oct 10 15:00:11] DEBUG[16429] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [httpstatus] len 10 [Oct 10 15:00:11] DEBUG[16429] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [phoneprov] len 9 [Oct 10 15:00:11] DEBUG[16429] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [static] len 6 [Oct 10 15:00:11] DEBUG[16429] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [ari] len 3 [Oct 10 15:00:11] DEBUG[16429] http.c: Match made with [ari] [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Finding handler for playback [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Checking endpoints [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Checking channels [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Checking events [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Checking recordings [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Checking playback [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Got it! [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Finding handler for 83fd10fa-6381-4ade-b55b-9427ac6a8065 [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Checking playbackId [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Got it! [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Finding handler for control [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Checking control [Oct 10 15:00:11] DEBUG[16429] res_ari.c: Got it! [Oct 10 15:00:12] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:13] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Difference is 8256, ms is 1052 [Oct 10 15:00:13] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Difference is 8256, ms is 1052 [Oct 10 15:00:13] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:17] DEBUG[16434] http.c: HTTP Request URI is /ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control?playbackid=83fd10fa-6381-4ade-b55b-9427ac6a8065&operation=restart&api_key=admin:secret [Oct 10 15:00:17] DEBUG[16434] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [httpstatus] len 10 [Oct 10 15:00:17] DEBUG[16434] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [phoneprov] len 9 [Oct 10 15:00:17] DEBUG[16434] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [static] len 6 [Oct 10 15:00:17] DEBUG[16434] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [ari] len 3 [Oct 10 15:00:17] DEBUG[16434] http.c: Match made with [ari] [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Finding handler for playback [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Checking endpoints [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Checking channels [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Checking events [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Checking recordings [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Checking playback [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Got it! [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Finding handler for 83fd10fa-6381-4ade-b55b-9427ac6a8065 [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Checking playbackId [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Got it! [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Finding handler for control [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Checking control [Oct 10 15:00:17] DEBUG[16434] res_ari.c: Got it! [Oct 10 15:00:17] DEBUG[16408][C-00000001] app.c: we'll restart the stream here at next loop [Oct 10 15:00:17] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 15:00:17] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 15:00:17] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format slin [Oct 10 15:00:17] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format gsm [Oct 10 15:00:17] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 15:00:17] VERBOSE[16408][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 15:00:18] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:18] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:23] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:23] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:26] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for QODzSz-932QKcSoWl6XMcphUiebV2olm - REGISTER (No RTP) [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 15:00:26] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_A [Oct 10 15:00:26] DEBUG[15569] chan_sip.c: Checking device state for peer phone_A [Oct 10 15:00:26] DEBUG[15569] devicestate.c: Changing state for SIP/phone_A - state 1 (Not in use) [Oct 10 15:00:26] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_A' [Oct 10 15:00:26] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_A' [Oct 10 15:00:26] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_A' [Oct 10 15:00:26] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_A' has not changed from 'Not in use' [Oct 10 15:00:26] DEBUG[15619] acl.c: For destination '10.24.19.97', our source address is '10.24.18.161'. [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for oRkcdmaaKAoFQidxqQmWjM-XX0FlRVOB - SUBSCRIBE (No RTP) [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_A" [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.19.97:5060 [Oct 10 15:00:26] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_A [Oct 10 15:00:26] DEBUG[15619] chan_sip.c: Destroying SIP dialog oRkcdmaaKAoFQidxqQmWjM-XX0FlRVOB [Oct 10 15:00:28] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:28] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:32] DEBUG[16436] http.c: HTTP Request URI is /ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control?playbackid=83fd10fa-6381-4ade-b55b-9427ac6a8065&operation&api_key=admin:secret [Oct 10 15:00:32] DEBUG[16436] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [httpstatus] len 10 [Oct 10 15:00:32] DEBUG[16436] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [phoneprov] len 9 [Oct 10 15:00:32] DEBUG[16436] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [static] len 6 [Oct 10 15:00:32] DEBUG[16436] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [ari] len 3 [Oct 10 15:00:32] DEBUG[16436] http.c: Match made with [ari] [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Finding handler for playback [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Checking endpoints [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Checking channels [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Checking events [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Checking recordings [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Checking playback [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Got it! [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Finding handler for 83fd10fa-6381-4ade-b55b-9427ac6a8065 [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Checking playbackId [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Got it! [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Finding handler for control [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Checking control [Oct 10 15:00:32] DEBUG[16436] res_ari.c: Got it! [Oct 10 15:00:33] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:33] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:34] DEBUG[16438] http.c: HTTP Request URI is /ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control?playbackid=83fd10fa-6381-4ade-b55b-9427ac6a8065&operation=restart&api_key=admin:secret [Oct 10 15:00:34] DEBUG[16438] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [httpstatus] len 10 [Oct 10 15:00:34] DEBUG[16438] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [phoneprov] len 9 [Oct 10 15:00:34] DEBUG[16438] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [static] len 6 [Oct 10 15:00:34] DEBUG[16438] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [ari] len 3 [Oct 10 15:00:34] DEBUG[16438] http.c: Match made with [ari] [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Finding handler for playback [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Checking endpoints [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Checking channels [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Checking events [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Checking recordings [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Checking playback [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Got it! [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Finding handler for 83fd10fa-6381-4ade-b55b-9427ac6a8065 [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Checking playbackId [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Got it! [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Finding handler for control [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Checking control [Oct 10 15:00:34] DEBUG[16438] res_ari.c: Got it! [Oct 10 15:00:34] DEBUG[16408][C-00000001] app.c: we'll restart the stream here at next loop [Oct 10 15:00:34] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 15:00:34] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Oct 10 15:00:34] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format slin [Oct 10 15:00:34] DEBUG[16408][C-00000001] channel.c: Set channel Announcer/ARI-00000025;1 to write format gsm [Oct 10 15:00:34] DEBUG[16408][C-00000001] channel.c: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Oct 10 15:00:34] VERBOSE[16408][C-00000001] file.c: -- Playing 'demo-congrats.gsm' (language 'en') [Oct 10 15:00:38] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:38] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:43] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:43] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:48] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for Nai21YMcL-HpkVmgs7Wk.v8zRDXvRSBI - REGISTER (No RTP) [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: **** Received REGISTER (2) - Command in SIP REGISTER [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: Store REGISTER's Contact header for call routing. [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: build_path: do not use Path headers [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 15:00:48] DEBUG[15569] devicestate.c: No provider found, checking channel drivers for SIP - phone_B [Oct 10 15:00:48] DEBUG[15569] chan_sip.c: Checking device state for peer phone_B [Oct 10 15:00:48] DEBUG[15569] devicestate.c: Changing state for SIP/phone_B - state 1 (Not in use) [Oct 10 15:00:48] DEBUG[15563] devicestate.c: Processing device state change for 'SIP/phone_B' [Oct 10 15:00:48] DEBUG[15563] devicestate.c: Adding per-server state of 'Not in use' for 'SIP/phone_B' [Oct 10 15:00:48] DEBUG[15563] devicestate.c: Aggregate devstate result is 'Not in use' for 'SIP/phone_B' [Oct 10 15:00:48] DEBUG[15563] devicestate.c: Aggregate state for device 'SIP/phone_B' has not changed from 'Not in use' [Oct 10 15:00:48] DEBUG[15619] acl.c: For destination '10.24.18.165', our source address is '10.24.18.161'. [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: Setting AST_TRANSPORT_UDP with address 10.24.18.161:5061 [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: Allocating new SIP dialog for ua47Mi1tsRrCcShK-Q33AxtB4NVNgAzE - SUBSCRIBE (No RTP) [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: **** Received SUBSCRIBE (10) - Command in SIP SUBSCRIBE [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: build_route: Contact hop: "phone_B" [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: Trying to put 'SIP/2.0 404' onto UDP socket destined for 10.24.18.165:5060 [Oct 10 15:00:48] NOTICE[15619] chan_sip.c: Received SIP subscribe for peer without mailbox: phone_B [Oct 10 15:00:48] DEBUG[15619] chan_sip.c: Destroying SIP dialog ua47Mi1tsRrCcShK-Q33AxtB4NVNgAzE [Oct 10 15:00:48] DEBUG[16086][C-00000001] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:48] DEBUG[16099][C-00000002] res_rtp_asterisk.c: Got RTCP report of 64 bytes [Oct 10 15:00:51] DEBUG[16441] http.c: HTTP Request URI is /ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control?playbackid=83fd10fa-6381-4ade-b55b-9427ac6a8065&api_key=admin:secret [Oct 10 15:00:51] DEBUG[16441] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [httpstatus] len 10 [Oct 10 15:00:51] DEBUG[16441] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [phoneprov] len 9 [Oct 10 15:00:51] DEBUG[16441] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [static] len 6 [Oct 10 15:00:51] DEBUG[16441] http.c: match request [ari/playback/83fd10fa-6381-4ade-b55b-9427ac6a8065/control] with handler [ari] len 3 [Oct 10 15:00:51] DEBUG[16441] http.c: Match made with [ari]